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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/trunk@137627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming 2008-08-13 22:33:32 +00:00
parent 81c3d0cfe4
commit 47ea5a01b4
2 changed files with 77 additions and 1 deletions
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26
UPGRADE.txt
View File

@ -84,6 +84,7 @@ 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
@ -92,6 +93,7 @@ Voicemail:
checking mailboxes for changes so that they can send MWI information to users.
Examples of situations that would require this option are web interfaces to
voicemail or an email client in the case of using IMAP storage.
* The externnotify script should accept an additional (last) parameter
containing the number of urgent messages in the INBOX.
@ -104,17 +106,22 @@ Applications:
been replaced with 'SUCCESS', and 'NOSUPPORT' has been replaced with
'UNSUPPORTED'). This change makes the SendImage application more consistent
with other 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
@ -125,20 +132,28 @@ 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.
* While app_directory has always relied on having a voicemail.conf or users.conf file
correctly set up, it now is dependent on app_voicemail being compiled as well.
* 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
@ -188,15 +203,18 @@ 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
The "username" setting still work, but is deprecated and will not work in
the next version of Asterisk.
* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
and you should start using that function instead for retrieving information about
the channel in a technology-agnostic way.
@ -204,6 +222,7 @@ 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.
@ -215,7 +234,12 @@ 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:

52
Zaptel-to-DAHDI.txt Normal file
View File

@ -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.