diff --git a/UPGRADE.txt b/UPGRADE.txt index 76e5577731..72f95c3e86 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -1,205 +1,10 @@ Information for Upgrading From Previous Asterisk Releases ========================================================= -Compiling: - -* The Asterisk 1.2 source code now uses C language features - supported only by 'modern' C compilers. Generally, this means GCC - version 3.0 or higher, although some GCC 2.96 releases will also - work. Some non-GCC compilers that support C99 and the common GCC - extensions (including anonymous structures and unions) will also - work. All releases of GCC 2.95 do _not_ have the requisite feature - support; systems using that compiler will need to be upgraded to - a more recent compiler release. - -Dialplan Expressions: - -* The dialplan expression parser (which handles $[ ... ] constructs) - has gone through a major upgrade, but has one incompatible change: - spaces are no longer required around expression operators, including - string comparisons. However, you can now use quoting to keep strings - together for comparison. For more details, please read the - doc/README.variables file, and check over your dialplan for possible - problems. - -Agents: - -* The default for ackcall has been changed to "no" instead of "yes" - because of a bug which caused the "yes" behavior to generally act like - "no". You may need to adjust the value if your agents behave - differently than you expect with respect to acknowledgement. - -* The AgentCallBackLogin application now requires a second '|' before - specifying an extension@context. This is to distinguish the options - string from the extension, so that they do not conflict. See - 'show application AgentCallbackLogin' for more details. - -Parking: - -* Parking behavior has changed slightly; when a parked call times out, - Asterisk will attempt to deliver the call back to the extension that - parked it, rather than the 's' extension. If that extension is busy - or unavailable, the parked call will be lost. - -Dialing: - -* The Caller*ID of the outbound leg is now the extension that was - called, rather than the Caller*ID of the inbound leg of the call. The - "o" flag for Dial can be used to restore the original behavior if - desired. Note that if you are looking for the originating callerid - from the manager event, there is a new manager event "Dial" which - provides the source and destination channels and callerid. - -IAX: - -* The naming convention for IAX channels has changed in two ways: - 1. The call number follows a "-" rather than a "/" character. - 2. The name of the channel has been simplified to IAX2/peer-callno, - rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno. - -SIP: - -* The global option "port" in 1.0.X that is used to set which port to - bind to has been changed to "bindport" to be more consistent with - the other channel drivers and to avoid confusion with the "port" - option for users/peers. - -* The "Registry" event now uses "Username" rather than "User" for - consistency with IAX. - Applications: -* With the addition of dialplan functions (which operate similarly - to variables), the SetVar application has been renamed to Set. - -* The CallerPres application has been removed. Use SetCallerPres - instead. It accepts both numeric and symbolic names. - -* The applications GetGroupCount, GetGroupMatchCount, SetGroup, and - CheckGroup have been deprecated in favor of functions. Here is a - table of their replacements: - - GetGroupCount([groupname][@category] GROUP_COUNT([groupname][@category]) Set(GROUPCOUNT=${GROUP_COUNT()}) - GroupMatchCount(groupmatch[@category]) GROUP_MATCH_COUNT(groupmatch[@category]) Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)}) - SetGroup(groupname[@category]) GROUP([category])=groupname Set(GROUP()=test) - CheckGroup(max[@category]) N/A GotoIf($[ ${GROUP_COUNT()} > 5 ]?103) - - Note that CheckGroup does not have a direct replacement. There is - also a new function called GROUP_LIST() which will return a space - separated list of all of the groups set on a channel. The GROUP() - function can also return the name of the group set on a channel when - used in a read environment. - -* The applications DBGet and DBPut have been deprecated in favor of - functions. Here is a table of their replacements: - - DBGet(foo=family/key) Set(foo=${DB(family/key)}) - DBPut(family/key=${foo}) Set(DB(family/key)=${foo}) - -* The application SetLanguage has been deprecated in favor of the - function LANGUAGE(). - - SetLanguage(fr) Set(LANGUAGE()=fr) - - The LANGUAGE function can also return the currently set language: - - Set(MYLANG=${LANGUAGE()}) - -* The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout - have been deprecated in favor of the function TIMEOUT(timeouttype): - - AbsoluteTimeout(300) Set(TIMEOUT(absolute)=300) - DigitTimeout(15) Set(TIMEOUT(digit)=15) - ResponseTimeout(15) Set(TIMEOUT(response)=15) - - The TIMEOUT() function can also return the currently set timeouts: - - Set(DTIMEOUT=${TIMEOUT(digit)}) - -* The applications SetCIDName, SetCIDNum, and SetRDNIS have been - deprecated in favor of the CALLERID(datatype) function: - - SetCIDName(Joe Cool) Set(CALLERID(name)=Joe Cool) - SetCIDNum(2025551212) Set(CALLERID(number)=2025551212) - SetRDNIS(2024561414) Set(CALLERID(RDNIS)=2024561414) - -* The application Record now uses the period to separate the filename - from the format, rather than the colon. - -* The application VoiceMail now supports a 'temporary' greeting for each - mailbox. This greeting can be recorded by using option 4 in the - 'mailbox options' menu, and 'change your password' option has been - moved to option 5. - -* The application VoiceMailMain now only matches the 'default' context if - none is specified in the arguments. (This was the previously - documented behavior, however, we didn't follow that behavior.) The old - behavior can be restored by setting searchcontexts=yes in voicemail.conf. - -Queues: - -* A queue is now considered empty not only if there are no members but if - none of the members are available (e.g. agents not logged on). To - restore the original behavior, use "leavewhenempty=strict" or - "joinwhenempty=strict" instead of "=yes" for those options. - -* It is now possible to use multi-digit extensions in the exit context - for a queue (although you should not have overlapping extensions, - as there is no digit timeout). This means that the EXITWITHKEY event - in queue_log can now contain a key field with more than a single - character in it. - -Extensions: - -* By default, there is a new option called "autofallthrough" in - extensions.conf that is set to yes. Asterisk 1.0 (and earlier) - behavior was to wait for an extension to be dialed after there were no - more extensions to execute. "autofallthrough" changes this behavior - so that the call will immediately be terminated with BUSY, - CONGESTION, or HANGUP based on Asterisk's best guess. If you are - writing an extension for IVR, you must use the WaitExten application - if "autofallthrough" is set to yes. - -AGI: - -* AGI scripts did not always get SIGHUP at the end, previously. That - behavior has been fixed. If you do not want your script to terminate - at the end of AGI being called (e.g. on a hangup) then set SIGHUP to - be ignored within your application. - -* CallerID is reported with agi_callerid and agi_calleridname instead - of a single parameter holding both. - -Music On Hold: - -* The preferred format for musiconhold.conf has changed; please see the - sample configuration file for the new format. The existing format - is still supported but will generate warnings when the module is loaded. - -chan_modem: - -* All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated - in this release, and will be removed in the next major Asterisk release. - Please migrate to chan_misdn for ISDN interfaces; there is no upgrade - path for aopen and bestdata modem users. - -MeetMe: - -* The conference application now allows users to increase/decrease their - speaking volume and listening volume (independently of each other and - other users); the 'admin' and 'user' menus have changed, and new sound - files are included with this release. However, if a user calling in - over a Zaptel channel that does NOT have hardware DTMF detection - increases their speaking volume, it is likely they will no longer be - able to enter/exit the menu or make any further adjustments, as the - software DTMF detector will not be able to recognize the DTMF coming - from their device. - -GetVar Manager Action: - -* Previously, the behavior of the GetVar manager action reported the value - of a variable in the following manner: - > name: value - This has been changed to a manner similar to the SetVar action and is now - > Variable: name - > Value: value +* In previous Asterisk releases, many applications would jump to priority n+101 + to indicate some kind of status or error condition. This functionality was + marked deprecated in Asterisk 1.2. An option to disable it was provided with + the default value set to 'on'. The default value for the global priority + jumping option is now 'off'. diff --git a/asterisk.c b/asterisk.c index 025b5aeea7..82c8466fe2 100644 --- a/asterisk.c +++ b/asterisk.c @@ -137,7 +137,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") */ /*! @{ */ -struct ast_flags ast_options = { AST_OPT_FLAG_TRANSCODE_VIA_SLIN | AST_OPT_FLAG_PRIORITY_JUMPING }; +struct ast_flags ast_options = { AST_OPT_FLAG_TRANSCODE_VIA_SLIN }; int option_verbose = 0; /*!< Verbosity level */ int option_debug = 0; /*!< Debug level */