===========================================================

===

=== Information for upgrading between Asterisk versions

===

=== These files document all the changes that MUST be taken

=== into account when upgrading between the Asterisk

=== versions listed below. These changes may require that

=== you modify your configuration files, dialplan or (in

=== some cases) source code if you have your own Asterisk

=== modules or patches. These files also includes advance

=== notice of any functionality that has been marked as

=== 'deprecated' and may be removed in a future release,

=== along with the suggested replacement functionality.

===

=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2

=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4

=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6

===

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From 1.8.13 to 1.8.14:

* permitdirectmedia/denydirectmedia now controls whether peers can be

bridged via directmedia by comparing the ACL to the bridging peer's

address rather than its own address.

From 1.8.12 to 1.8.13:

* The complex processor detection and optimization has been removed from

the makefile in favor of using native optimization suppport when available.

BUILD_NATIVE can be disabled via menuselect under "Compiler Flags".

From 1.8.10 to 1.8.11:

* If no transport is specified in sip.conf, transport will default to UDP.

Also, if multiple transport= lines are used, only the last will be used.

From 1.6.2 to 1.8:

* chan_sip no longer sets HASH(SIP_CAUSE,<chan name>) on channels by default.

This must now be enabled by setting 'sipstorecause' to 'yes' in sip.conf.

This carries a performance penalty.

* Asterisk now requires libpri 1.4.11+ for PRI support.

* A couple of CLI commands in res_ais were changed back to their original form:

"ais show clm members" --> "ais clm show members"

"ais show evt event channels" --> "ais evt show event channels"

* The default value for 'autofill' and 'shared_lastcall' in queues.conf has

been changed to 'yes'.

* The default value for the alwaysauthreject option in sip.conf has been changed

from "no" to "yes".

* The behavior of the 'parkedcallstimeout' has changed slightly. The formulation

of the extension name that a timed out parked call is delivered to when this

option is set to 'no' was modified such that instead of converting '/' to '0',

the '/' is converted to an underscore '_'. See the updated documentation in

features.conf.sample for more information on the behavior of the

'parkedcallstimeout' option.

* Asterisk-addons no longer exists as an independent package. Those modules

now live in the addons directory of the main Asterisk source tree. They

are not enabled by default. For more information about why modules live in

addons, see README-addons.txt.

* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few

users of this channel in the tree have been converted to LOG_NOTICE or removed

(in cases where the same message was already generated to another channel).

* The usage of RTP inside of Asterisk has now become modularized. This means

the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.

If you are not using autoload=yes in modules.conf you will need to ensure

it is set to load. If not, then any module which uses RTP (such as chan_sip)

will not be able to send or receive calls.

* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still

remains. It now exists within app_chanspy.c and retains the exact same

functionality as before.

* The default behavior for Set, AGI, and pbx_realtime has been changed to implement

1.6 behavior by default, if there is no [compat] section in asterisk.conf. In

prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades.

Specifically, that means that pbx_realtime and res_agi expect you to use commas

to separate arguments in applications, and Set only takes a single pair of

a variable name/value. The old 1.4 behavior may still be obtained by setting

app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of

asterisk.conf.

* The PRI channels in chan_dahdi can no longer change the channel name if a

different B channel is selected during call negotiation. To prevent using

the channel name to infer what B channel a call is using and to avoid name

collisions, the channel name format is changed.

The new channel naming for PRI channels is:

DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>

* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type)

so the dialplan can determine the B channel currently in use by the channel.

Use CHANNEL(no_media_path) to determine if the channel even has a B channel.

* Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk

channel so AMI applications can passively determine the B channel currently

in use. Calls with "no-media" as the DAHDIChannel do not have an associated

B channel. No-media calls are either on hold or call-waiting.

* The ChanIsAvail application has been changed so the AVAILSTATUS variable

no longer contains both the device state and cause code. The cause code

is now available in the AVAILCAUSECODE variable. If existing dialplan logic

is written to expect AVAILSTATUS to contain the cause code it needs to be

changed to use AVAILCAUSECODE.

* ExternalIVR will now send Z events for invalid or missing files, T events

now include the interrupted file and bugs in argument parsing have been

fixed so there may be arguments specified in incorrect ways that were

working that will no longer work. Please see

https://wiki.asterisk.org/wiki/display/AST/External+IVR+Interface for details.

* OSP lookup application changes following variable names:

OSPPEERIP to OSPINPEERIP

OSPTECH to OSPOUTTECH

OSPDEST to OSPDESTINATION

OSPCALLING to OSPOUTCALLING

OSPCALLED to OSPOUTCALLED

OSPRESULTS to OSPDESTREMAILS

* The Manager event 'iax2 show peers' output has been updated. It now has a

similar output of 'sip show peers'.

* VoiceMailMain and VMAuthenticate, if a '*' is entered in the first position

of a Mailbox or Password, will, if it exists, jump to the 'a' extension in

the current dialplan context.

* The CALLERPRES() dialplan function is deprecated in favor of

CALLERID(num-pres) and CALLERID(name-pres).

* Environment variables that start with "AST_" are reserved to the system and

may no longer be set from the dialplan.

* When a call is redirected inside of a Dial, the app and appdata fields of the

CDR will now be set to "AppDial" and "(Outgoing Line)" instead of being blank.

* The CDR handling of billsec and duration field has changed. If your table

definition specifies those fields as float,double or similar they will now

be logged with microsecond accuracy instead of a whole integer.

* chan_sip will no longer set up a local call forward when receiving a

482 Loop Detected response. The dialplan will just continue from where it

left off.

* The 'stunaddr' option has been removed from chan_sip. This feature did not

behave as expected, had no correct use case, and was not RFC compliant. The

removal of this feature will hopefully be followed by a correct RFC compliant

STUN implementation in chan_sip in the future.

* The default value for the pedantic option in sip.conf has been changed

from "no" to "yes".

* The ConnectedLineNum and ConnectedLineName headers were added to many AMI

events/responses if the CallerIDNum/CallerIDName headers were also present.

The addition of connected line support changes the behavior of the channel

caller ID somewhat. The channel caller ID value no longer time shares with

the connected line ID on outgoing call legs. The timing of some AMI

events/responses output the connected line ID as caller ID. These party ID's

are now separate.

* The Dial application d and H options do not automatically answer the call

anymore. It broke DTMF attended transfers. Since many SIP and ISDN phones

cannot send DTMF before a call is connected, you need to answer the call

leg to those phones before using Dial with these options for them to have

any effect before the dialed party answers.

* The outgoing directory (where .call files are read) now uses inotify to

detect file changes instead of polling the directory on a regular basis.

If your outgoing folder is on a NFS mount or another network file system,

changes to the files will not be detected. You can revert to polling the

directory by specifying --without-inotify to configure before compiling.

* The 'sipusers' realtime table has been removed completely. Use the 'sippeers'

table with type 'user' for user type objects.

* The sip.conf allowoverlap option now accepts 'dtmf' as a value. If you

are using the early media DTMF overlap dialing method you now need to set

allowoverlap=dtmf.

From 1.6.1 to 1.6.2:

* SIP no longer sends the 183 progress message for early media by

default. Applications requiring early media should use the

progress() dialplan app to generate the progress message.

* The firmware for the IAXy has been removed from Asterisk. It can be

downloaded from http://downloads.digium.com/pub/iaxy/. To have Asterisk

install the firmware into its proper location, place the firmware in the

contrib/firmware/iax/ directory in the Asterisk source tree before running

"make install".

* T.38 FAX error correction mode can no longer be configured in udptl.conf;

instead, it is configured on a per-peer (or global) basis in sip.conf, with

the same default as was present in udptl.conf.sample.

* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;

instead, it is either supplied by the application servicing the T.38 channel

(for a FAX send or receive) or calculated from the bridged endpoint's

maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf

allows for overriding the value supplied by a remote endpoint, which is useful

when T.38 connections are made to gateways that supply incorrectly-calculated

maximum datagram sizes.

* There have been some changes to the IAX2 protocol to address the security

concerns documented in the security advisory AST-2009-006. Please see the

IAX2 security document, doc/IAX2-security.pdf, for information regarding

backwards compatibility with versions of Asterisk that do not contain these

changes to IAX2.

* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers

has been renamed to 'directmedia', to better reflect what it actually does.

In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,

starting and stopping music-on-hold, and other reasons, and the 'canreinvite'

option never had any effect on these cases, it only affected the re-INVITEs

used for direct media path setup. For MGCP and Skinny, the option was poorly

named because those protocols don't even use INVITE messages at all. For

backwards compatibility, the old option is still supported in both normal

and Realtime configuration files, but all of the sample configuration files,

Realtime/LDAP schemas, and other documentation refer to it using the new name.

* The default console now will use colors according to the default background

color, instead of forcing the background color to black. If you are using a

light colored background for your console, you may wish to use the option

flag '-W' to present better color choices for the various messages. However,

if you'd prefer the old method of forcing colors to white text on a black

background, the compatibility option -B is provided for this purpose.

* SendImage() no longer hangs up the channel on transmission error or on

any other error; in those cases, a FAILURE status is stored in

SENDIMAGESTATUS and dialplan execution continues. The possible

return values stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and

UNSUPPORTED. ('OK' has been replaced with 'SUCCESS', and 'NOSUPPORT'

has been replaced with 'UNSUPPORTED'). This change makes the

SendImage application more consistent with other applications.

* skinny.conf now has separate sections for lines and devices.

Please have a look at configs/skinny.conf.sample and update

your skinny.conf.

* Queue names previously were treated in a case-sensitive manner,

meaning that queues with names like "sales" and "sALeS" would be

seen as unique queues. The parsing logic has changed to use

case-insensitive comparisons now when originally hashing based on

queue names, meaning that now the two queues mentioned as examples

earlier will be seen as having the same name.

* The SPRINTF() dialplan function has been moved into its own module,

func_sprintf, and is no longer included in func_strings. If you use this

function and do not use 'autoload=yes' in modules.conf, you will need

to explicitly load func_sprintf for it to be available.

* The res_indications module has been removed. Its functionality was important

enough that most of it has been moved into the Asterisk core.

Two applications previously provided by res_indications, PlayTones and

StopPlayTones, have been moved into a new module, app_playtones.

* Support for Taiwanese was incorrectly supported with the "tw" language code.

In reality, the "tw" language code is reserved for the Twi language, native

to Ghana. If you were previously using the "tw" language code, you should

switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan

specific localizations. Additionally, "mx" should be changed to "es_MX",

Georgian was incorrectly specified as "ge" but should be "ka", and Czech is

"cs", not "cz".

* DAHDISendCallreroutingFacility() parameters are now comma-separated,

instead of the old pipe.

* res_jabber: autoprune has been disabled by default, to avoid misconfiguration

that would end up being interpreted as a bug once Asterisk started removing

the contacts from a user list.

* The cdr.conf file must exist and be configured correctly in order for CDR

records to be written.

* cdr_pgsql now assumes the encoding of strings it is handed are in LATIN9,

which should cover most uses of the extended ASCII set. If your strings

use a different encoding in Asterisk, the "encoding" parameter may be set

to specify the correct character set.

From 1.6.0.1 to 1.6.1:

* The ast_agi_register_multiple() and ast_agi_unregister_multiple()

API calls were added in 1.6.0, so that modules that provide multiple

AGI commands could register/unregister them all with a single

step. However, these API calls were not implemented properly, and did

not allow the caller to know whether registration or unregistration

succeeded or failed. They have been redefined to now return success

or failure, but this means any code using these functions will need

be recompiled after upgrading to a version of Asterisk containing

these changes. In addition, the source code using these functions

should be reviewed to ensure it can properly react to failure

of registration or unregistration of its API commands.

* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()

to better match what it really does, and the argument order has been

changed to be consistent with other API calls that perform similar

operations.

From 1.6.0.x to 1.6.1:

* In previous versions of Asterisk, due to the way objects were arranged in

memory by chan_sip, the order of entries in sip.conf could be adjusted to

control the behavior of matching against peers and users. The way objects

are managed has been significantly changed for reasons involving performance

and stability. A side effect of these changes is that the order of entries

in sip.conf can no longer be relied upon to control behavior.

* The following core commands dealing with dialplan have been deprecated: 'core

show globals', 'core set global' and 'core set chanvar'. Use the equivalent

'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'

instead.

* In the dialplan expression parser, the logical value of spaces

immediately preceding a standalone 0 previously evaluated to

true. It now evaluates to false. This has confused a good many

people in the past (typically because they failed to realize the

space had any significance). Since this violates the Principle of

Least Surprise, it has been changed.

* While app_directory has always relied on having a voicemail.conf or users.conf file

correctly set up, it now is dependent on app_voicemail being compiled as well.

* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),

and you should start using that function instead for retrieving information about

the channel in a technology-agnostic way.

* If you have any third party modules which use a config file variable whose

name ends in a '+', please note that the append capability added to this

version may now conflict with that variable naming scheme. An easy

workaround is to ensure that a space occurs between the '+' and the '=',

to differentiate your variable from the append operator. This potential

conflict is unlikely, but is documented here to be thorough.

* The "Join" event from app_queue now uses the CallerIDNum header instead of

the CallerID header to indicate the CallerID number.

* If you use ODBC storage for voicemail, there is a new field called "flag"

which should be a char(8) or larger. This field specifies whether or not a

message has been designated to be "Urgent", "PRIORITY", or not.