2008-11-19 13:03:50 +00:00
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===========================================================
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2008-11-21 20:43:43 +00:00
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===
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2008-11-19 13:03:50 +00:00
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=== Information for upgrading between Asterisk 1.6 versions
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2008-02-08 16:49:19 +00:00
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===
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2008-11-21 20:43:43 +00:00
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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 includes 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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2008-02-08 16:49:19 +00:00
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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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2008-11-19 13:03:50 +00:00
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=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
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2008-11-21 20:43:43 +00:00
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===
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2008-11-19 13:03:50 +00:00
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===========================================================
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2009-09-16 13:37:32 +00:00
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As of 1.6.1.7:
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* The firmware for the IAXy has been removed from Asterisk. It can be
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downloaded from http://downloads.digium.com/pub/iaxy/. To have Asterisk
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install the firmware into its proper location, place the firmware in the
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contrib/firmware/iax/ directory in the Asterisk source tree before running
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"make install".
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2009-10-01 16:19:43 +00:00
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* T.38 FAX error correction mode can no longer be configured in udptl.conf;
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instead, it is configured on a per-peer (or global) basis in sip.conf, with
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the same default as was present in udptl.conf.sample.
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2009-10-05 19:53:18 +00:00
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* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
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instead, it is either supplied by the application servicing the T.38 channel
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(for a FAX send or receive) or calculated from the bridged endpoint's
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maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
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allows for overriding the value supplied by a remote endpoint, which is useful
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when T.38 connections are made to gateways that supply incorrectly-calculated
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maximum datagram sizes.
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2009-09-03 19:42:59 +00:00
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As of 1.6.1.6:
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2009-09-03 19:42:00 +00:00
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* There have been some changes to the IAX2 protocol to address the security
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concerns documented in the security advisory AST-2009-006. Please see the
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IAX2 security document, doc/IAX2-security.pdf, for information regarding
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backwards compatibility with versions of Asterisk that do not contain these
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changes to IAX2.
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Merged revisions 208464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 22:21:57 +00:00
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From 1.6.1.1 to 1.6.1.2:
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2009-07-23 22:32:17 +00:00
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* Beginning with this release, Asterisk's internal methods of
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negotiating T.38 (FAX over IP) sessions changed in
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non-backwards-compatible ways. Any applications that previously used
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AST_CONTROL_T38 control frames will have to be upgraded to use
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AST_CONTROL_T38_PARAMETERS control frames instead; app_fax.c is a good
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example of how to generate and respond to these frames. These changes
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were made to solve significant T.38 interoperability problems between
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Asterisk and various SIP/T.38 endpoints identified by many users of
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Asterisk.
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Merged revisions 208464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 22:21:57 +00:00
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2008-11-19 13:03:50 +00:00
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From 1.6.0.1 to 1.6.1:
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* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
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API calls were added in 1.6.0, so that modules that provide multiple
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AGI commands could register/unregister them all with a single
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step. However, these API calls were not implemented properly, and did
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not allow the caller to know whether registration or unregistration
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succeeded or failed. They have been redefined to now return success
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or failure, but this means any code using these functions will need
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be recompiled after upgrading to a version of Asterisk containing
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these changes. In addition, the source code using these functions
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should be reviewed to ensure it can properly react to failure
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of registration or unregistration of its API commands.
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* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
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to better match what it really does, and the argument order has been
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changed to be consistent with other API calls that perform similar
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operations.
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From 1.6.0.x to 1.6.1:
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2009-02-27 21:25:10 +00:00
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* In previous versions of Asterisk, due to the way objects were arranged in
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memory by chan_sip, the order of entries in sip.conf could be adjusted to
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control the behavior of matching against peers and users. The way objects
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are managed has been significantly changed for reasons involving performance
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and stability. A side effect of these changes is that the order of entries
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in sip.conf can no longer be relied upon to control behavior.
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2008-11-19 13:03:50 +00:00
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* The following core commands dealing with dialplan have been deprecated: 'core
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2008-07-17 14:00:27 +00:00
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show globals', 'core set global' and 'core set chanvar'. Use the equivalent
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'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
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instead.
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2008-11-19 13:03:50 +00:00
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* In the dialplan expression parser, the logical value of spaces
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immediately preceding a standalone 0 previously evaluated to
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true. It now evaluates to false. This has confused a good many
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people in the past (typically because they failed to realize the
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space had any significance). Since this violates the Principle of
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Least Surprise, it has been changed.
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2008-08-13 22:33:32 +00:00
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2008-04-29 18:48:26 +00:00
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* While app_directory has always relied on having a voicemail.conf or users.conf file
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correctly set up, it now is dependent on app_voicemail being compiled as well.
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2008-08-13 22:33:32 +00:00
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2008-06-15 15:21:16 +00:00
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* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
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and you should start using that function instead for retrieving information about
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the channel in a technology-agnostic way.
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2007-05-04 16:37:23 +00:00
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2008-08-05 18:25:16 +00:00
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* If you have any third party modules which use a config file variable whose
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name ends in a '+', please note that the append capability added to this
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version may now conflict with that variable naming scheme. An easy
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workaround is to ensure that a space occurs between the '+' and the '=',
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to differentiate your variable from the append operator. This potential
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conflict is unlikely, but is documented here to be thorough.
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2008-11-25 00:41:07 +00:00
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* The "Join" event from app_queue now uses the CallerIDNum header instead of
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the CallerID header to indicate the CallerID number.
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2009-06-30 18:44:26 +00:00
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* Support for Taiwanese was incorrectly supported with the "tw" language code.
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In reality, the "tw" language code is reserved for the Twi language, native
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to Ghana. If you were previously using the "tw" language code, you should
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switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
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2009-06-30 21:30:32 +00:00
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specific localizations. Additionally, "mx" should be changed to "es_MX",
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Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
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"cs", not "cz".
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2009-06-30 18:44:26 +00:00
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2009-07-17 22:11:35 +00:00
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* If you use ODBC storage for voicemail, there is a new field called "flag"
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which should be a char(8) or larger. This field specifies whether or not a
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message has been designated to be "Urgent", "PRIORITY", or not.
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2009-06-30 18:44:26 +00:00
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From 1.6.1.1 to 1.6.1.2:
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* Support for Taiwanese was incorrectly supported with the "tw" language code.
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In reality, the "tw" language code is reserved for the Twi language, native
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to Ghana. If you were previously using the "tw" language code, you should
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switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
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2009-06-30 21:30:32 +00:00
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specific localizations. Additionally, "mx" should be changed to "es_MX",
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Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
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"cs", not "cz".
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2009-06-30 18:44:26 +00:00
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