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	Merged revisions 137627 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ 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 ........ 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@137631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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							| @@ -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: | ||||
|   | ||||
							
								
								
									
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							| @@ -0,0 +1,52 @@ | ||||
| ========================================================= | ||||
| === Information for upgrading from Zaptel to DAHDI    === | ||||
| ========================================================= | ||||
|  | ||||
| As announced in early 2008, Digium is renaming the Zaptel telephony | ||||
| interface project to DAHDI (Digium Asterisk Hardware Device Interface) | ||||
| to accommodate the desires of the owner of the Zaptel trademark for | ||||
| telephony purposes. | ||||
|  | ||||
| This version of Asterisk can only be built using DAHDI, and as a | ||||
| result there are number of changes that will be visible to previous | ||||
| users of Asterisk with Zaptel. | ||||
|  | ||||
| First, the modules that directly use services from DAHDI have been | ||||
| renamed; the new names are: | ||||
|  | ||||
|   chan_zap.so -> chan_dahdi.so | ||||
|   app_zapbarge.so -> app_dahdibarge.so | ||||
|   app_zapras.so -> app_dahdiras.so | ||||
|   app_zapscan.so -> app_dahdiscan.so | ||||
|  | ||||
| Second, the behavior of many modules has changed due to the switch to | ||||
| DAHDI; the changes are listed below. | ||||
|  | ||||
| chan_dahdi.so: | ||||
|  | ||||
|   Incoming and outgoing channels managed by this module will be | ||||
|   'DAHDI' channels instead of 'Zap' channels. | ||||
|  | ||||
|   All CLI commands that began with 'zap' have been renamed to 'dahdi' | ||||
|   commands. | ||||
|  | ||||
|   All Asterisk Manager Interface (AMI) actions that began with 'Zap' | ||||
|   have changed to 'DAHDI' prefixes. | ||||
|  | ||||
|   The ZapSendKeypadFacility dialplan application has been renamed to | ||||
|   DAHDISendKeypadFacility. | ||||
|  | ||||
|   The configuration for the channel driver will be read from | ||||
|   /etc/asterisk/chan_dahdi.conf instead of /etc/asterisk/zapata.conf. | ||||
|  | ||||
| app_dahdibarge.so: | ||||
|  | ||||
|   The ZapBarge application has been renamed to DAHDIBarge. | ||||
|  | ||||
| app_dahdiras.so: | ||||
|  | ||||
|   The ZapRAS application has been renamed to DAHDIRAS. | ||||
|  | ||||
| app_dahdiscan.so: | ||||
|  | ||||
|   The ZapScan application has been renamed to DAHDIScan. | ||||
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