The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
258 lines
13 KiB
Text
258 lines
13 KiB
Text
===========================================================
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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 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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===
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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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===
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===========================================================
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From 1.6.2 to 1.8:
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* Asterisk-addons no longer exists as an independent package. Those modules
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now live in the addons directory of the main Asterisk source tree. They
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are not enabled by default. For more information about why modules live in
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addons, see README-addons.txt.
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* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
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users of this channel in the tree have been converted to LOG_NOTICE or removed
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(in cases where the same message was already generated to another channel).
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* The usage of RTP inside of Asterisk has now become modularized. This means
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the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
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If you are not using autoload=yes in modules.conf you will need to ensure
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it is set to load. If not, then any module which uses RTP (such as chan_sip)
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will not be able to send or receive calls.
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* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still
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remains. It now exists within app_chanspy.c and retains the exact same
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functionality as before.
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* The default behavior for Set, AGI, and pbx_realtime has been changed to implement
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1.6 behavior by default, if there is no [compat] section in asterisk.conf. In
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prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades.
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Specifically, that means that pbx_realtime and res_agi expect you to use commas
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to separate arguments in applications, and Set only takes a single pair of
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a variable name/value. The old 1.4 behavior may still be obtained by setting
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app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of
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asterisk.conf.
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* The PRI channels in chan_dahdi can no longer change the channel name if a
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different B channel is selected during call negotiation. To prevent using
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the channel name to infer what B channel a call is using and to avoid name
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collisions, the channel name format is changed.
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The new channel naming for PRI channels is:
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DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
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* The ChanIsAvail application has been changed so the AVAILSTATUS variable
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no longer contains both the device state and cause code. The cause code
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is now available in the AVAILCAUSECODE variable. If existing dialplan logic
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is written to expect AVAILSTATUS to contain the cause code it needs to be
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changed to use AVAILCAUSECODE.
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* ExternalIVR will now send Z events for invalid or missing files, T events
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now include the interrupted file and bugs in argument parsing have been
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fixed so there may be arguments specified in incorrect ways that were
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working that will no longer work.
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Please see doc/externalivr.txt for details.
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* OSP lookup application changes following variable names:
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OSPPEERIP to OSPINPEERIP
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OSPTECH to OSPOUTTECH
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OSPDEST to OSPDESTINATION
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OSPCALLING to OSPOUTCALLING
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OSPCALLED to OSPOUTCALLED
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OSPRESULTS to OSPDESTREMAILS
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* The Manager event 'iax2 show peers' output has been updated. It now has a
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similar output of 'sip show peers'.
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* VoiceMailMain and VMAuthenticate, if a '*' is entered in the first position
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of a Mailbox or Password, will, if it exists, jump to the 'a' extension in
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the current dialplan context.
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* The CALLERPRES() dialplan function is deprecated in favor of
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CALLERID(num-pres) and CALLERID(name-pres).
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* Environment variables that start with "AST_" are reserved to the system and
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may no longer be set from the dialplan.
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* When a call is redirected inside of a Dial, the app and appdata fields of the
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CDR will now be set to "AppDial" and "(Outgoing Line)" instead of being blank.
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* The CDR handling of billsec and duration field has changed. If your table
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definition specifies those fields as float,double or similar they will now
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be logged with microsecond accuracy instead of a whole integer.
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* chan_sip will no longer set up a local call forward when receiving a
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482 Loop Detected response. The dialplan will just continue from where it
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left off.
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From 1.6.1 to 1.6.2:
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* SIP no longer sends the 183 progress message for early media by
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default. Applications requiring early media should use the
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progress() dialplan app to generate the progress message.
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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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* 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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* 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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* 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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* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
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has been renamed to 'directmedia', to better reflect what it actually does.
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In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
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starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
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option never had any effect on these cases, it only affected the re-INVITEs
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used for direct media path setup. For MGCP and Skinny, the option was poorly
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named because those protocols don't even use INVITE messages at all. For
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backwards compatibility, the old option is still supported in both normal
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and Realtime configuration files, but all of the sample configuration files,
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Realtime/LDAP schemas, and other documentation refer to it using the new name.
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* The default console now will use colors according to the default background
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color, instead of forcing the background color to black. If you are using a
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light colored background for your console, you may wish to use the option
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flag '-W' to present better color choices for the various messages. However,
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if you'd prefer the old method of forcing colors to white text on a black
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background, the compatibility option -B is provided for this purpose.
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* SendImage() no longer hangs up the channel on transmission error or on
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any other error; in those cases, a FAILURE status is stored in
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SENDIMAGESTATUS and dialplan execution continues. The possible
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return values stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and
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UNSUPPORTED. ('OK' has been replaced with 'SUCCESS', and 'NOSUPPORT'
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has been replaced with 'UNSUPPORTED'). This change makes the
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SendImage application more consistent with other applications.
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* skinny.conf now has separate sections for lines and devices.
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Please have a look at configs/skinny.conf.sample and update
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your skinny.conf.
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* Queue names previously were treated in a case-sensitive manner,
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meaning that queues with names like "sales" and "sALeS" would be
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seen as unique queues. The parsing logic has changed to use
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case-insensitive comparisons now when originally hashing based on
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queue names, meaning that now the two queues mentioned as examples
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earlier will be seen as having the same name.
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* The SPRINTF() dialplan function has been moved into its own module,
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func_sprintf, and is no longer included in func_strings. If you use this
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function and do not use 'autoload=yes' in modules.conf, you will need
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to explicitly load func_sprintf for it to be available.
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* The res_indications module has been removed. Its functionality was important
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enough that most of it has been moved into the Asterisk core.
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Two applications previously provided by res_indications, PlayTones and
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StopPlayTones, have been moved into a new module, app_playtones.
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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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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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* DAHDISendCallreroutingFacility() parameters are now comma-separated,
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instead of the old pipe.
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* res_jabber: autoprune has been disabled by default, to avoid misconfiguration
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that would end up being interpreted as a bug once Asterisk started removing
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the contacts from a user list.
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* The cdr.conf file must exist and be configured correctly in order for CDR
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records to be written.
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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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* 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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* The following core commands dealing with dialplan have been deprecated: 'core
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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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* 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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* 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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* 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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* 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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* 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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* 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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