Asterisk v1.2 upgrade to v1.4 gotchas

The upgrade is fairly smooth, and there's lots of new features. It's the deprecated stuff that bites you during the migration process.

Start the migration with reading UPGRADE.txt, and then look at the CHANGES file for details that were introduced with 1.4.0.

Make sure to read the new xxx.conf.sample files. That way you may detect new features/options that not seldomly also fix potential security issues.

asterisk.conf

For sure you will want to have "internal_timing=yes"!

extensions.conf

Hurray, you may now monitor the call park and Meet conference with hint, use "Meetme:1234" or "park:701@parkedcalls"!
Call pickup has changed, in particular you really must take a look at PICKUPMARK.

A line starting with ;-- (semicolon immediately followed by two dashes) is now treated as opening a multi-line comment, so be aware! You might disable the entirety of what is remaining in your dialplan from this point on.

Changes to watch out for:
  • Calling a voicemail box with flags for busy or unavailable (options b and u) must now be performed with a pipe as opposed to prepending that option to the mailbox number: "b1234" or "u4567" turns into "1234|b" and "4567|u"
  • SIP_HEADER() with (Via) now needs to be written as (Via,1) in Asterisk 1.4
  • LookupCIDName is deprecated. Please use the much more beautiful and easy-to-read Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) instead. Note that this must now typically be combined with a conditional statement like ExecIf() if you want to keep the current CallerID name in case the AstDb does not have (better) information on this caller.

Many similar changes for variables are described in ugprade.txt:
  • change ${TIMESTAMP} variable to ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} function
  • change ${CALLERIDNUM} variable to ${CALLERID(num)} function


sip.conf

  • In the [general] section "port=" has been renamed to "bindport=" to prevent misunderstandings
  • The default for QoS settings has changed from the old TOS to the new DiffServ method. This also applies to iax.conf, by the way.
  • with the new subscribemwi=yes we can finally instruct Asterisk to not send what some SIP devices consider as unsolicited NOTIFY messages (AVM Fritz!Box, Siemens Gigaset and others). This prevents SIP ERROR 481 or "Remote host cannot match NOTIFY"

BLF and hints

  • you will need to set "call-limit=" to make hints (SIP SUBSCRIPTIONS) work in Asterisk 1.4
  • also look at the general setting "limitonpeers=yes" and "notifyringing=yes" etc.



iax.conf

The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up.

  • in general section, add: iaxthreadcount = 200
  • in general section, add: iaxmaxthreadcount = 1000

Later in 1.4.26.2 also this changed due to a security issue:

add this to iax.conf: calltokenoptional = 0.0.0.0/0.0.0.0
add this to the [guest] user in iax.conf: requirecalltoken=no (many guests will be using old Asterisk boxes)
In future: Upgrade the IAX peers and provide call tokens!

zaptel turns into dahdi

During the summer 2008 and after the release of 1.4.17 (?) zaptel has been renamed to dahdi. Since zaptel/dahdi provide timing to MeetMe this also matters for users that do not have any zapte (Digium) hardware (ztdummy vs. dahdi_dummy). Also the zaphfc module for the HFC-S ISDN cards is affected. See UPGRADE.txt in your dahdi-linux source directory for details how to migrate.

Asterisk 1.4 will continue to have support for Zaptel, although it will be enhanced to also transparently support DAHDI instead, and the documentation (and default configuration files) will encourage new users to use DAHDI instead of Zaptel.

AgentCallBackLogin()

AgentCallBackLogin is deprecated. If you were using this functionality, it can be replaced using dialplan logic, but there appears to be no way to update hints (if you use them) dynamically.

Music on Hold

Digium has strongly advised everyone in Aug. 2009 to discard the previously shipped music-on-hold files due to potential license issues.

Incompatibilities

The g726 codec mess

Look at "g726nonstandard=yes" and "allow=g726aal2" - this is mentioned in upgrade.txt, but maybe you have missed to understand that you will need to review every single case where you have g726 enabled. In particular make sure you test cases where Asterisk is supposed to transcode between call legs with standard g726 and g726nonstandard!

Obtaining and compiling the ilbc codec

The download script in contrib/ forgets to copy the ilbc code to asterisk/codecs/ilbc

g723 removed

codec_g723 was removed

rfc2833compensate

In Asterisk 1.4 you will need to turn on rfc2833compensate if you want to cleanly interact on DTMF with an Asterisk 1.2 system.

Patches that would have deserved to be in 1.4

See http://users.netplex.net/~andrew/asterisk/, and in particular:
  1. Codec Negotiation mod
  2. Remote CID update mod
  3. G.722 Codec mod
  4. FAX app (spandsp)


Official Information for Upgrading From Previous Asterisk Releases


Build Process (configure script):


Asterisk now uses an autoconf-generated configuration script to learn how it
should build itself for your system. As it is a standard script, running:

$ ./configure --help

will show you all the options available. This script can be used to tell the
build process what libraries you have on your system (if it cannot find them
automatically), which libraries you wish to have ignored even though they may
be present, etc.

You must run the configure script before Asterisk will build, although it will
attempt to automatically run it for you with no options specified; for most
users, that will result in a similar build to what they would have had before
the configure script was added to the build process (except for having to run
'make' again after the configure script is run). Note that the configure script
does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
when your system configuration changes or you wish to build Asterisk with
different options.

Build Process (module selection):


The Asterisk source tree now includes a basic module selection and build option
selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
In this tool, you can disable building of modules that you don't care about,
turn on/off global options for the build and see which modules will not
(and cannot) be built because your system does not have the required external
dependencies installed.

The resulting file from menuselect is called 'menuselect.makeopts'. Note that
the resulting menuselect.makeopts file generally contains which modules *not*
to build. The modules listed in this file indicate which modules have unmet
dependencies, a present conflict, or have been disabled by the user in the
menuselect interface. Compiler Flags can also be set in the menuselect
interface. In this case, the resulting file contains which CFLAGS are in use,
not which ones are not in use.

If you would like to save your choices and have them applied against all
builds, the file can be copied to '~/.asterisk.makeopts' or
'/etc/asterisk.makeopts'.

Build Process (Makefile targets):


The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
in the menuselect tool.

It is now possible to run most make targets against a single subdirectory; from
the top level directory, for example, 'make channels' will run 'make all' in the
'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.

Sound (prompt) and Music On Hold files:


Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
use with Asterisk have been replaced with new versions produced from high quality
master recordings, and are available in three languages (English, French and
Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
In addition, the music on hold files provided by FreePlay Music are now available
in the same five formats, but no longer available in MP3 format.

The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
(as were supplied with previous releases) and the FreePlay MOH files in WAV format.
All of the other variations can be installed by running 'make menuselect' and
selecting the packages you wish to install; when you run 'make install', those
packages will be downloaded and installed along with the standard files included
in the tarball.

If for some reason you expect to not have Internet access at the time you will be
running 'make install', you can make your package selections using menuselect and
then run 'make sounds' to download (only) the sound packages; this will leave the
sound packages in the 'sounds' subdirectory to be used later during installation.

WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
instead of the alternate-language files being stored in subdirectories underneath
the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
language itself, then places all the sound files for that language under that
directory and its subdirectories. This is the layout that will be created if you
select non-English languages to be installed via menuselect, HOWEVER Asterisk does
not default to this layout and will not find the files in the places it expects them
to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
/etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
installed.

PBX Core:


  • The (very old and undocumented) ability to use BYEXTENSION for dialing
instead of ${EXTEN} has been removed.

  • Builtin (res_features) transfer functionality attempts to use the context
defined in TRANSFER_CONTEXT variable of the transferer channel first. If
not set, it uses the transferee variable. If not set in any channel, it will
attempt to use the last non macro context. If not possible, it will default
to the current context.

  • The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
if your dialplan relies on the ability to 'run off the end' of an extension
and wait for a new extension without using WaitExten() to accomplish that,
you will need set autofallthrough to 'no' in your extensions.conf file.

Command Line Interface:


  • 'show channels concise', designed to be used by applications that will parse
its output, previously used ':' characters to separate fields. However, some
of those fields can easily contain that character, making the output not
parseable. The delimiter has been changed to '!'.

Applications:


  • 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'.

  • The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
been removed in this version. You should use the equivalent dialplan
function in places where you have previously used one of these applications.

  • The application SetGlobalVar has been deprecated. You should replace uses
of this application with the following combination of Set and GLOBAL():
Set(GLOBAL(name)=value). You may also access global variables exclusively by
using the GLOBAL() dialplan function, instead of relying on variable
interpolation falling back to globals when no channel variable is set.

  • The application SetVar has been renamed to Set. The syntax SetVar was marked
deprecated in version 1.2 and is no longer recognized in this version.

  • app_read has been updated to use the newer options codes, using "skip" or
"noanswer" will not work. Use s or n. Also there is a new feature i, for
using indication tones, so typing in skip would give you unexpected results.

  • OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.

  • The CONNECT event in the queue_log from app_queue now has a second field
in addition to the holdtime field. It contains the unique ID of the
queue member channel that is taking the call. This is useful when trying
to link recording filenames back to a particular call from the queue.

  • The old/current behavior of app_queue has a serial type behavior
in that the queue will make all waiting callers wait in the queue
even if there is more than one available member ready to take
calls until the head caller is connected with the member they
were trying to get to. The next waiting caller in line then
becomes the head caller, and they are then connected with the
next available member and all available members and waiting callers
waits while this happens. This cycle continues until there are
no more available members or waiting callers, whichever comes first.
The new behavior, enabled by setting autofill=yes in queues.conf
either at the general level to default for all queues or
to set on a per-queue level, makes sure that when the waiting
callers are connecting with available members in a parallel fashion
until there are no more available members or no more waiting callers,
whichever comes first. This is probably more along the lines of how
one would expect a queue should work and in most cases, you will want
to enable this new behavior. If you do not specify or comment out this
option, it will default to "no" to keep backward compatability with the old
behavior.

  • Queues depend on the channel driver reporting the proper state
for each member of the queue. To get proper signalling on
queue members that use the SIP channel driver, you need to
enable a call limit (could be set to a high value so it
is not put into action) and also make sure that both inbound
and outbound calls are accounted for.

Example:

        [general]
        limitonpeers = yes
 
        [peername]
        type=friend
        call-limit=10
 


  • The app_queue application now has the ability to use MixMonitor to
record conversations queue members are having with queue callers. Please
see configs/queues.conf.sample for more information on this option.

  • The app_queue application strategy called 'roundrobin' has been deprecated
for this release. Users are encouraged to use 'rrmemory' instead, since it
provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
'rrmemory' will be renamed 'roundrobin'.

  • app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
the 'm' option now provides the functionality of "initially muted".
In practice, most existing dialplans using the 'm' flag should not notice
any difference, unless the keypad menu is enabled, allowing the user
to unmute themsleves.

  • ast_play_and_record would attempt to cancel the recording if a DTMF
'0' was received. This behavior was not documented in most of the
applications that used ast_play_and_record and the return codes from
ast_play_and_record weren't checked for properly.
ast_play_and_record has been changed so that '0' no longer cancels a
recording. If you want to allow DTMF digits to cancel an
in-progress recording use ast_play_and_record_full which allows you
to specify which DTMF digits can be used to accept a recording and
which digits can be used to cancel a recording.

  • ast_app_messagecount has been renamed to ast_app_inboxcount. There is now a
new ast_app_messagecount function which takes a single context/mailbox/folder
mailbox specification and returns the message count for that folder only.
This addresses the deficiency of not being able to count the number of
messages in folders other than INBOX and Old.

  • The exit behavior of the AGI applications has changed. Previously, when
a connection to an AGI server failed, the application would cause the channel
to immediately stop dialplan execution and hangup. Now, the only time that
the AGI applications will cause the channel to stop dialplan execution is
when the channel itself requests hangup. The AGI applications now set an
AGISTATUS variable which will allow you to find out whether running the AGI
was successful or not.

Previously, there was no way to handle the case where Asterisk was unable to
locally execute an AGI script for some reason. In this case, dialplan
execution will continue as it did before, but the AGISTATUS variable will be
set to "FAILURE".

A locally executed AGI script can now exit with a non-zero exit code and this
failure will be detected by Asterisk. If an AGI script exits with a non-zero
exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
"SUCCESS".

  • app_voicemail: The ODBC_STORAGE capability now requires the extended table format
previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
your table format using the schema provided in doc/odbcstorage.txt

  • app_waitforsilence: Fixes have been made to this application which changes the
default behavior with how quickly it returns. You can maintain "old-style" behavior
with the addition/use of a third "timeout" parameter.
Please consult the application documentation and make changes to your dialplan
if appropriate.

Manager:


  • After executing the 'status' manager action, the "Status" manager events
included the header "CallerID:" which was actually only the CallerID number,
and not the full CallerID string. This header has been renamed to
"CallerIDNum". For compatibility purposes, the CallerID parameter will remain
until after the release of 1.4, when it will be removed. Please use the time
during the 1.4 release to make this transition.

  • The AgentConnect event now has an additional field called "BridgedChannel"
which contains the unique ID of the queue member channel that is taking the
call. This is useful when trying to link recording filenames back to
a particular call from the queue.

  • app_userevent has been modified to always send Event: UserEvent with the
additional header UserEvent: <userspec>. Also, the Channel and UniqueID
headers are not automatically sent, unless you specify them as separate
arguments. Please see the application help for the new syntax.

  • app_meetme: Mute and Unmute events are now reported via the Manager API.
Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
are easier to use than "Action Command:". The MeetMeStopTalking event has
also been deprecated in favor of the already existing MeetmeTalking event
with a "Status" of "on" or "off" added.

  • OriginateFailure and OriginateSuccess events were replaced by event
OriginateResponse with a header named "Response" to indicate success or
failure

Variables:


  • The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
and ${LANGUAGE} have all been deprecated in favor of their related dialplan
functions. You are encouraged to move towards the associated dialplan
function, as these variables will be removed in a future release.

  • The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
adjustable from cdr.conf, instead of recompiling.

  • OSP applications exports several new variables, ${OSPINHANDLE},
${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}

  • Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
created channel. This variables holds the channel name of the transferer.

  • The dial plan variable PRI_CAUSE will be removed from future versions
of Asterisk.
It is replaced by adding a cause value to the hangup() application.

Functions:


  • The function ${CHECK_MD5()} has been deprecated in favor of using an
expression: $[${MD5(<string>)} = ${saved_md5}].

  • The 'builtin' functions that used to be combined in pbx_functions.so are
now built as separate modules. If you are not using 'autoload=yes' in your
modules.conf file then you will need to explicitly load the modules that
contain the functions you want to use.

  • The ENUMLOOKUP() function with the 'c' option (for counting the number of
records), but the lookup fails to match any records, the returned value will
now be "0" instead of blank.

  • The REALTIME() function is now available in version 1.4 and app_realtime has
been deprecated in favor of the new function. app_realtime will be removed
completely with the version 1.6 release so please take the time between
releases to make any necessary changes

  • The QUEUEAGENTCOUNT() function has been deprecated in favor of
QUEUE_MEMBER_COUNT().

The IAX2 channel:


  • The "mailboxdetail" option has been deprecated. Previously, if this option
was not enabled, the 2 byte MSGCOUNT information element would be set to all
1's to indicate there there is some number of messages waiting. With this
option enabled, the number of new messages were placed in one byte and the
number of old messages are placed in the other. This is now the default
(and the only) behavior.

The SIP channel:


  • The "incominglimit" setting is replaced by the "call-limit" setting in
sip.conf.

  • OSP support code is removed from SIP channel to OSP applications. ospauth
option in sip.conf is removed to osp.conf as authpolicy. allowguest option
in sip.conf cannot be set as osp anymore.

  • The Asterisk RTP stack has been changed in regards to RFC2833 reception
and transmission. Packets will now be sent with proper duration instead of all
at once. If you are receiving calls from a pre-1.4 Asterisk installation you
will want to turn on the rfc2833compensate option. Without this option your
DTMF reception may act poorly.

  • The $SIPUSERAGENT dialplan variable is deprecated and will be removed
in coming versions of Asterisk. Please use the dialplan function
SIPCHANINFO(useragent) instead.

  • The ALERT_INFO dialplan variable is deprecated and will be removed
in coming versions of Asterisk. Please use the dialplan application
sipaddheader() to add the "Alert-Info" header to the outbound invite.

  • The "canreinvite" option has changed. canreinvite=yes used to disable
re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat
to disable re-invites when NAT=yes. This is propably what you want.
The settings are now: "yes", "no", "nonat", "update". Please consult
sip.conf.sample for detailed information.

The Zap channel:


  • Support for MFC/R2 has been removed, as it has not been functional for some
time and it has no maintainer.

The Agent channel:


  • Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
it provided can be done using dialplan logic, without requiring additional
channel and module locks (which frequently caused deadlocks). An example of
how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.

The G726-32 codec:


  • It has been determined that previous versions of Asterisk used the wrong codeword
packing order for G726-32 data. This version supports both available packing orders,
and can transcode between them. It also now selects the proper order when
negotiating with a SIP peer based on the codec name supplied in the SDP. However,
there are existing devices that improperly request one order and then use another;
Sipura and Grandstream ATAs are known to do this, and there may be others. To
be able to continue to use these devices with this version of Asterisk and the
G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
to sip.conf, so that Asterisk can use the packing order expected by the device (even

though it requested a different order). In addition, the internal format number for
G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
result of this is that this version of Asterisk will be able to interoperate over
IAX2 with older versions of Asterisk, as long as this version is told to allow
'g726aal2' instead of 'g726' as the codec for the call.

Installation:


  • On BSD systems, the installation directories have changed to more "FreeBSDish"
directories. On startup, Asterisk will look for the main configuration in
/usr/local/etc/asterisk/asterisk.conf
If you have an old installation, you might want to remove the binaries and
move the configuration files to the new locations. The following directories
are now default:
ASTLIBDIR /usr/local/lib/asterisk
ASTVARLIBDIR /usr/local/share/asterisk
ASTETCDIR /usr/local/etc/asterisk
ASTBINDIR /usr/local/bin/asterisk
ASTSBINDIR /usr/local/sbin/asterisk

Music on Hold:


  • The music on hold handling has been changed in some significant ways in hopes
to make it work in a way that is much less confusing to users. Behavior will
not change if the same configuration is used from older versions of Asterisk.
However, there are some new configuration options that will make things work
in a way that makes more sense.

Previously, many of the channel drivers had an option called "musicclass" or
something similar. This option set what music on hold class this channel
would *hear* when put on hold. Some people expected (with good reason) that
this option was to configure what music on hold class to play when putting
the bridged channel on hold. This option has now been deprecated.

Two new music on hold related configuration options for channel drivers have
been introduced. Some channel drivers support both options, some just one,
and some support neither of them. Check the sample configuration files to see
which options apply to which channel driver.

The "mohsuggest" option specifies which music on hold class to suggest to the
bridged channel when putting them on hold. The only way that this class can
be overridden is if the bridged channel has a specific music class set that
was done in the dialplan using Set(CHANNEL(musicclass)=something).

The "mohinterpret" option is similar to the old "musicclass" option. It
specifies which music on hold class this channel would like to listen to when
put on hold. This music class is only effective if this channel has no music
class set on it from the dialplan and the bridged channel putting this one on
hold had no "mohsuggest" setting.

The IAX2 and Zap channel drivers have an additional feature for the
"mohinterpret" option. If this option is set to "passthrough", then these
channel drivers will pass through the HOLD message in signalling instead of
starting music on hold on the channel. An example for how this would be
useful is in an enterprise network of Asterisk servers. When one phone on one
server puts a phone on a different server on hold, the remote server will be
responsible for playing the hold music to its local phone that was put on
hold instead of the far end server across the network playing the music.

CDR Records:


  • The behavior of the "clid" field of the CDR has always been that it will
contain the callerid ANI if it is set, or the callerid number if ANI was not
set. When using the "callerid" option for various channel drivers, some
would set ANI and some would not. This has been cleared up so that all
channel drivers set ANI. If you would like to change the callerid number
on the channel from the dialplan and have that change also show up in the
CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).

API:


  • There are some API functions that were not previously prefixed with the 'ast_'
prefix but now are; these include the ADSI, ODBC and AGI interfaces. If you
have a module that uses the services provided by res_adsi, res_odbc, or
res_agi, you will need to add ast_ prefixes to the functions that you call
from those modules.

Formats:


  • format_wav: The GAIN preprocessor definition has been changed from 2 to 0
in Asterisk 1.4. This change was made in response to user complaints of
choppiness or the clipping of loud signal peaks. The GAIN preprocessor
definition will be retained in Asterisk 1.4, but will be removed in a
future release. The use of GAIN for the increasing of voicemail message
volume should use the 'volgain' option in voicemail.conf


Created by: cmaj, Last modification: Mon 21 of May, 2012 (22:37 UTC) by admin


Please update this page with new information, just login and click on the "Edit" or "Discussion" tab. Get a free login here: Register Thanks! - Find us on Google+

Page Changes | Comments

 

Featured -

Search: