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author | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-08-13 22:34:25 +0000 |
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committer | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-08-13 22:34:25 +0000 |
commit | 1e8b08cb2538177a939b40dfb2bf90cf5a1c702b (patch) | |
tree | d1883e89718401ca751a3572e47c43fec7ebea28 /UPGRADE.txt | |
parent | 9466c30453ea62aba29a6dcb1434f0d00358db83 (diff) |
Merged revisions 137627 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug 2008) | 9 lines
Merged revisions 137530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug 2008) | 1 line
add document describing what users will need to be aware of when upgrading to this version and using DAHDI
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@137631 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'UPGRADE.txt')
-rw-r--r-- | UPGRADE.txt | 23 |
1 files changed, 22 insertions, 1 deletions
diff --git a/UPGRADE.txt b/UPGRADE.txt index 5e0f6e88e..8fb0a0f71 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -70,6 +70,7 @@ Voicemail: to make them more distinguishable from 'maxmsgs', which sets folder size. The old variables will continue to work in this version, albeit with a deprecation warning. + * If you use any interface for modifying voicemail aside from the built in dialplan applications, then the option "pollmailboxes" *must* be set in voicemail.conf for message waiting indication (MWI) to work properly. This @@ -81,17 +82,22 @@ Voicemail: Applications: + * ChanIsAvail() now has a 't' option, which allows the specified device to be queried for state without consulting the channel drivers. This performs mostly a 'ChanExists' sort of function. + * ChannelRedirect() will not terminate the channel that fails to do a channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS will reflect if the attempt was successful of not. + * SetCallerPres() has been replaced with the CALLERPRES() dialplan function and is now deprecated. + * DISA()'s fifth argument is now an options argument. If you have previously used 'NOANSWER' in this argument, you'll need to convert that to the new option 'n'. + * Macro() is now deprecated. If you need subroutines, you should use the Gosub()/Return() applications. To replace MacroExclusive(), we have introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK(). You may use @@ -102,18 +108,25 @@ Applications: sake of backwards compatibility it will not be removed . It is also worth noting that using both Macro() and GoSub() at the same time is _heavily_ discouraged. + * Read() now sets a READSTATUS variable on exit. It does NOT automatically return -1 (and hangup) anymore on error. If you want to hangup on error, you need to do so explicitly in your dialplan. + * Privacy() no longer uses privacy.conf, so any options must be specified directly in the application arguments. + * MusicOnHold application now has duration parameter which allows specifying timeout in seconds. + * WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold. + * SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...) instead. + * The arguments in ExecIf changed a bit, to be more like other applications. The syntax is now ExecIf(<cond>?appiftrue(args):appiffalse(args)). + * The behavior of the Set application now depends upon a compatibility option, set in asterisk.conf. To use the old 1.4 behavior, which allowed Set to take multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf. To @@ -163,10 +176,12 @@ Channel Drivers: file names and formats are all controlled via the normal mechanisms. If the user has not configured the automon feature, the normal "415 Unsupported media type" is returned, and nothing is done. + * SIP: The "call-limit" option is marked as deprecated. It still works in this version of Asterisk, but will be removed in the following version. Please use the groupcount functions in the dialplan to enforce call limits. The "limitonpeer" configuration option is now renamed to "counteronpeer". + * SIP: The "username" option is now renamed to "defaultuser" to match "defaultip". These are used only before registration to call a peer with the uri sip:defaultuser@defaultip @@ -176,6 +191,7 @@ Channel Drivers: * chan_local.c: the comma delimiter inside the channel name has been changed to a semicolon, in order to make the Local channel driver compatible with the comma delimiter change in applications. + * H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio" to be compatible with settings in sip.conf. The "tos" and "cos" configuration is deprecated and will stop working in the next release of Asterisk. @@ -187,7 +203,12 @@ Channel Drivers: to modify modules.conf to add another "noload" line to ensure that only one of these three modules gets loaded. -* Zap: The "msdstrip" option has been deprecated, as it provides no value over +* DAHDI: The chan_zap module that supported PSTN interfaces using + Zaptel has been renamed to chan_dahdi, and only supports the DAHDI + telephony driver package for PSTN interfaces. See the + Zaptel-to-DAHDI.txt file for more details on this transition. + +* DAHDI: The "msdstrip" option has been deprecated, as it provides no value over the method of stripping digits in the dialplan using variable substring syntax. Configuration: |