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標題: 2013/01/03 Announcement Of Latest Asterisk Release- 1.8.19.0, 10.11.0, 11.1.2 [打印本頁]

作者: ckleea 時間: 2010-8-11 06:10 標題: 2013/01/03 Announcement Of Latest Asterisk Release- 1.8.19.0, 10.11.0, 11.1.2

本帖最後由 ckleea 於 2013-1-4 06:31 編輯

The Asterisk Development Team has announced the release of Asterisk 1.6.2.11.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.11 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are a few of the issues resolved by community developers:

* Send DialPlanComplete as a response, not as a separate event. Otherwise, it
goes to all manager sessions and may exclude the current session, if the
Events mask excludes it.
(Closes issue #17504. Reported, patched by rrb3942)

* Allow the "useragent" value to be restored into memory from the realtime
backend. This value is purely informational. It does not alter configuration
at all.
(Closes issue #16029. Reported, patched by Guggemand)

* Fix rt(c)p set debug ip taking wrong argument Also clean up some coding
errors.
(Closes issue #17469. Reported, patched by wdoekes)

* Ensure channel placed in meetme in ringing state is properly hung up. An
outgoing channel placed in meetme while still ringing which was then hung up
would not exit meetme and the channel was not properly destroyed.
(Closes issue #15871. Reported, patched by Ivan)

* Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers.
(Closes issue #16102. Reported, patched by Delvar)

* cdr_pgsql does not detect when a table is found. This change adds an ERROR
message to let you know when a failure exists to get the columns from the
pgsql database, which typically means that the table does not exist.
(Closes issue #17478. Reported, patched by kobaz)

* Avoid crashing when installing a duplicate translation path with a lower
cost.
(Closes issue #17092. Reported, patched by moy)

* Add missing handling for ringing state for use with queue empty options.
(Closes issue #17471. Reported, patched by jazzy)

* Fix reporting estimated queue hold time. Just say the number of seconds
(after minutes) rather than doing some incorrect calculation with respect to
minutes.
(Closes issue #17498. Reported, patched by corruptor)

For a full list of changes in the current release, please see the
ChangeLog:

http://downloads.asterisk.org/pu ... /ChangeLog-1.6.2.11

Thank you for your continued support of Asterisk!

--

作者: ckleea 時間: 2010-8-11 06:10

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta3.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any issues found to the issue tracker,
http://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4. For more information about
support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

This release contains fixes since the last beta release as reported by the
community. A sampling of the changes in this release include:

* Fix a regression where HTTP would always be enabled regardless of setting.
(Closes issue #17708. Reported, patched by pabelanger)

* ACL errors displayed on screen when using dynamic_exclude_static in sip.conf
(Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson)

* Support "channels" in addition to "channel" in chan_dahdi.conf.
(https://reviewboard.asterisk.org/r/804)

* Fix parsing error in sip_sipredirect(). The code was written in a way that
did a bad job of parsing the port out of a URI. Specifically, it would do
badly when dealing with an IPv6 address.
(Closes issue #17661. Reported by oej. Patched by mmichelson)

* Fix inband DTMF detection on outgoing ISDN calls.
(Patched by russellb and rmudgett)

* Fixes issue with translator frame not getting freed. This issue prevented
g729 licenses from being freed up.
(Closes issue #17630. Reported by manvirr. Patched by dvossel)

* Fixed IPv6-related SIP parsing bugs and updated documention.
(Reported by oej. Patched by sperreault)

* Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a
list of a specified item. Matches up with FIELDQTY() and CUT().
(Closes #17713. Reported, patched by gareth. Tested by tilghman)

Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:

* Secure RTP
* IPv6 Support
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/aster ... ANGES?view=checkout

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... angeLog-1.8.0-beta3

Thank you for your continued support of Asterisk!

作者: ckleea 時間: 2010-8-11 06:11

The Asterisk Development Team has announced the release of Asterisk 1.4.35.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.35 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are a few of the issues resolved by community developers:

* Ensure channel placed in meetme in ringing state is properly hung up.
(Closes issue #15871. Reported, patched by Ivan)

* If all members are paused, the wrong status is indicated.
(Closes issue #17576. Reported, patched by ramonpeek)

* Fix logging message for stale nonce.
(Closes issue #17582. Reported, patched by kenner)

* Interpret device state AST_DEVICE_UNKNOWN as extension state
AST_EXTENSION_NOT_INUSE.
(Closes issue #16035. Reported by francesco_r. Patched by viniciusfontes)

* Resolve T.38 negotiation regression.
(Closes issue #16705. Reported by mpiazzatnetbug. Patched by ebroad)

For a full list of changes in the current release, please see the
ChangeLog:

http://downloads.asterisk.org/pu ... sk/ChangeLog-1.4.35

Thank you for your continued support of Asterisk!

作者: 角色 時間: 2010-8-11 08:56

Thank you for your information. If the Asterisk 1.8 is a stable version rather than beta version, I shall install it without anydoubt.

YH

作者: ckleea 時間: 2010-8-11 09:15

Same here for 1.8

But 1.6.2.11 rpm is not yet ready to update

作者: bubblestar 時間: 2010-8-11 11:06

Thanks for the information.

作者: ckleea 時間: 2010-8-18 07:22

Centos RPM Asterisk 1.6.2.11 just available for download

I have made a service restart to see if there is any difference.

作者: 角色 時間: 2010-8-18 07:38

I have already updated the Asterisk 1.4.35. As recommened by ckleea CHing, Asterisk upgrade using yum is very convenient.

YH

作者: ckleea 時間: 2010-8-27 08:07

有趣？剛剛yum update again asterisk 1.6.11。唔知點解？

作者: 角色 時間: 2010-8-28 17:54

不明白你上面的意思！那么能否用yum update了？

角色

作者: ckleea 時間: 2010-8-29 06:09

update once before but somehow I can update again.

作者: 角色 時間: 2010-8-29 06:51

Asterisk 1.8 beta 4 has already been released. Once it is in production version, I shall consider to install it.

YH

作者: ckleea 時間: 2010-9-4 22:26

YH, This may be useful for you

The Asterisk Development Team is pleased to announce the release of
DAHDI-Linux and DAHDI-Tools version 2.4.0.

DAHDI-Linux 2.4.0, DAHDI-Tools 2.4.0, and DAHDI-Linux-Complete are
available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

In addition to several bug fixes, the most significant changes from the
2.3.0 release are:

General DAHDI Changes:

* Added DAHDI_MAINT_ALARM_SIM maintenance mode for drivers that
support alarm simulation (wct4xxp).  This is only used by
dahdi_maint and doesn't change the ABI.

* Span callbacks are moved out of the dahdi_span structure potentially
saving memory when a single driver implements multiple spans.

Updated Drivers:

* wctdm24xxp, wcte12xp: Fix bug when moving to memory mapped registers
where the interrupt handler was run twice for every interrupt.

* wctdm24xxp, wcte12xp: Processing moved back to interrupt handler.
(Closes issue #17289 Reported by alecdavis)

* wctdm24xxp, wcte12xp: Update VPMADT032 firmware to 1.25.  Contains
improvements to prevent loss of convergence when signal levels go
over a certain threshold and for handling line condition changes.

* wctdm24xxp: Fix race conditions/improvements in FXS line feed register
handling.
(Closes issues #17724 and #17764. Reported and patched by alecdavis)

* wctdm24xxp: Added "companding" module parameter to replace
"alawoverride".  When BRI modules are installed on a Hx8 board alaw is
the default companding so change the semantics to just allow the
companding to be forced as opposed to overriding a default.  The
default is "auto" which means alaw if there are BRI modules, otherwise
ulaw.

* wctdm24xxp: Set 'spantype' for digital spans so that they can be
displayed with dahdi_scan.

* wcte12xp: dahdi_cfg does not need to be called twice when using RBS
signalling.

* wcte12xp: Loopback module parameter removed since 'dahdi_maint' can
now put the spans in digital loopback.

* wct4xxp: Add 'latency', 'max_latency', and 'ms_per_irq' module
parameters to set expected latency conditions when using Gen5
firmware.

* wct4xxp: Added support for network loopback modes via dahdi_maint.

* wct4xxp: Which span is providing card timing is now exported via
sysfs.

* wcb4xxp: Fixed pulse mask for improved TBR3 compliance.

* wcb4xxp: Added pci-ids for Junghanns PCI-E cards.

* wcb4xxp: Added 'companding' module parameter.

* wcb4xxp: Fixed bug when using automatic timing sync.

* wcb4xxp: Which span is providing card timing is now exported via
sysfs.

* wctdm: Added configurable debounce to support old rotary phones.
(Closes issue #16339.  Reported by alecdavis patch by tilghman.)

* xpp:
FXS: support VMWI config from Asterisk >= 1.6.1
  PRI:
- PRI Astribanks always sync AB (and independent)
- don't send "duplicates" in E1 as in D4
- Reduce noise at E1 startup.
- T1 CAS fixes.
PIC 4 rev. 7381: fix T1 returning signaling register in non-CAS

Changes to dahdi-tools:

* dahdi_maint: Added support for simulating alarm conditions.

* dahdi_scan: Report more detailed alarm information.

* xpp_fxloader:
- Load firmware in the background
- Support 1163 twinstar devices
- A delay loop for older kernels (e.g. 2.6.18)

* astribank_is_starting does not depend on libusb.

* Allow using CONNECTOR/LABEL in genconf_parameters for pri_termtype

For a full list of changes in these releases, please see the ChangeLogs at
http://svn.asterisk.org/svn/dahdi/linux/tags/2.4.0/ChangeLog and
http://svn.asterisk.org/svn/dahdi/tools/tags/2.4.0/ChangeLog

Issues found in these releases can be reported at http://issues.asterisk.org

Thank you for your continued support of Asterisk!

作者: 角色 時間: 2010-9-5 07:34

Copied. Thanks.

YH

作者: ckleea 時間: 2010-9-16 21:14

The Asterisk Development Team has announced the release of Asterisk 1.4.36. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.36 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

* Fix issue where DNID does not get cleared on a new call when using
   immediate=yes with ISDN signaling.
   (Closes issue #17568. Reported by wuwu. Patched by rmudgett)
* Fix issue where SIP promiscuous redirect could fail to dial the
   redirect (app_queue).
* Fixes issue with translator frame not getting freed. This issue prevented
   G.729 licenses from being freed up.
   (Closes issue #17630. Reported by manvirr. Patched by dvossel)
* Ensure SSRC is changed when media source is changed to resolve audio delay.
   (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
* Q931 - Sending PROGRESS after sending ALERTING is a protocol error.
   (Closes issue #17874. Reported, patched by nic_bellamy)

For a full list of changes in the current release, please see the
ChangeLog:

http://downloads.asterisk.org/pu ... sk/ChangeLog-1.4.36

Thank you for your continued support of Asterisk!

作者: 角色 時間: 2010-9-16 21:18

Just installed a few hours ago.

YH

作者: ckleea 時間: 2010-9-16 21:18

The Asterisk Development Team has announced the release of Asterisk 1.6.2.13.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

This release resolves an issue where the .version and ChangeLog files were not
updated for 1.6.2.12. Asterisk 1.6.2.13 has no additional changes from 1.6.2.12
other than the .version, ChangeLog and summary files.

For a full list of changes in the current release, please see the
ChangeLog:

http://downloads.asterisk.org/pu ... /ChangeLog-1.6.2.13

Thank you for your continued support of Asterisk!

I just upgrade to 1.6.2.13

作者: Sam 時間: 2010-9-17 11:17

Just complied 1.6.2.13 in my Buffalo ARM NAS, have to remove 1.4 first and try it later today.

作者: 角色 時間: 2010-9-17 11:23

Where did you get the compilear for your ARM NAS?

YH

作者: ckleea 時間: 2010-9-17 13:04

回復 18# Sam

Do you have problem in installation? I try 1.6 for dd-wrt before but somehow not working.

作者: Sam 時間: 2010-9-17 14:07

Where did you get the compilear for your ARM NAS?

YH
角色發表於 2010-9-17 11:23

2 methods:
1) Codesourcery has created the cross toolchain for ARM Processors, you can install this toolchain in Ubuntu. 3 different version of Codesourcery Cross ToolChain can be downloaded here

2) Install GCC it in the NAS box.

I use option 2, a bit slow but it is a more easy way.

作者: Sam 時間: 2010-9-17 14:08

回復 Sam

Do you have problem in installation? I try 1.6 for dd-wrt before but somehow not working ...
ckleea 發表於 2010-9-17 13:04

Still in progress... I will report the result later.

作者: Sam 時間: 2010-9-17 15:25

Ok, seems working but command syntax seems different from v1.4:

# /opt/usr/sbin/asterisk -vvvgc
u-Law coding table test complete.
u-Law tandem transcoding test complete.
a-Law coding tables test complete.
a-Law tandem transcoding test complete.
Asterisk 1.6.2.13, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Started Asterisk Event Logger
No 'modules.conf' found, no modules will be loaded.
Asterisk Event Logger Started /var/log/asterisk/event_log
Asterisk Dynamic Loader Starting:
  == Manager registered action Ping
Unable to open AMI configuration manager.conf, or configuration is invalid. Asterisk management interface (AMI) disabled.
  == Manager registered action Events
Couldn't find function EXCEPTION in XML documentation
Couldn't find function EXCEPTION in XML documentation
Couldn't find application Answer in XML documentation
Couldn't find application Answer in XML documentation
Couldn't find application BackGround in XML documentation
Couldn't find application BackGround in XML documentation
Couldn't find application Busy in XML documentation
Couldn't find application Busy in XML documentation
Couldn't find application Congestion in XML documentation
Couldn't find application Congestion in XML documentation
Couldn't find application ExecIfTime in XML documentation
Couldn't find application ExecIfTime in XML documentation
Couldn't find application Goto in XML documentation
Couldn't find application Goto in XML documentation
Couldn't find application GotoIf in XML documentation
Couldn't find application GotoIf in XML documentation
Couldn't find application GotoIfTime in XML documentation
Couldn't find application GotoIfTime in XML documentation
Couldn't find application ImportVar in XML documentation
Couldn't find application ImportVar in XML documentation
Couldn't find application Hangup in XML documentation
Couldn't find application Hangup in XML documentation
Couldn't find application Incomplete in XML documentation
Couldn't find application Incomplete in XML documentation
Couldn't find application NoOp in XML documentation
Couldn't find application NoOp in XML documentation
Couldn't find application Proceeding in XML documentation
Couldn't find application Proceeding in XML documentation
Couldn't find application Progress in XML documentation
Couldn't find application Progress in XML documentation
Couldn't find application RaiseException in XML documentation
Couldn't find application RaiseException in XML documentation
Couldn't find application ResetCDR in XML documentation
Couldn't find application ResetCDR in XML documentation
Couldn't find application Ringing in XML documentation
Couldn't find application Ringing in XML documentation
Couldn't find application SayAlpha in XML documentation
Couldn't find application SayAlpha in XML documentation
Couldn't find application SayDigits in XML documentation
Couldn't find application SayDigits in XML documentation
Couldn't find application SayNumber in XML documentation
Couldn't find application SayNumber in XML documentation
Couldn't find application SayPhonetic in XML documentation
Couldn't find application SayPhonetic in XML documentation
Couldn't find application Set in XML documentation
Couldn't find application Set in XML documentation
Couldn't find application MSet in XML documentation
Couldn't find application MSet in XML documentation
Couldn't find application SetAMAFlags in XML documentation
Couldn't find application SetAMAFlags in XML documentation
Couldn't find application Wait in XML documentation
Couldn't find application Wait in XML documentation
Couldn't find application WaitExten in XML documentation
Couldn't find application WaitExten in XML documentation
Can't find indications config file indications.conf.
Couldn't find application Bridge in XML documentation
Couldn't find application Bridge in XML documentation
Could not load features.conf
Couldn't find application ParkedCall in XML documentation
Couldn't find application ParkedCall in XML documentation
Couldn't find application Park in XML documentation
Couldn't find application Park in XML documentation
Unable to open Asterisk database '/var/lib/asterisk/astdb': No such file or directory
No 'modules.conf' found, no modules will be loaded.
*CLI> == Manager registered action Logoff
  == Manager registered action Login
  == Manager registered action Challenge
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action GetConfig
  == Manager registered action GetConfigJSON
  == Manager registered action UpdateConfig
  == Manager registered action CreateConfig
  == Manager registered action ListCategories
  == Manager registered action Redirect
  == Manager registered action Atxfer
  == Manager registered action Originate
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxStatus
  == Manager registered action MailboxCount
  == Manager registered action ListCommands
  == Manager registered action SendText
  == Manager registered action UserEvent
  == Manager registered action WaitEvent
  == Manager registered action CoreSettings
  == Manager registered action CoreStatus
  == Manager registered action Reload
  == Manager registered action CoreShowChannels
  == Manager registered action ModuleLoad
  == Manager registered action ModuleCheck
Asterisk PBX Core Initializing
Registering builtin applications:
  == Registered custom function 'EXCEPTION'
[Answer]
  == Registered application 'Answer'
[BackGround]
  == Registered application 'BackGround'
[Busy]
  == Registered application 'Busy'
[Congestion]
  == Registered application 'Congestion'
[ExecIfTime]
  == Registered application 'ExecIfTime'
[Goto]
  == Registered application 'Goto'
[GotoIf]
  == Registered application 'GotoIf'
[GotoIfTime]
  == Registered application 'GotoIfTime'
[ImportVar]
  == Registered application 'ImportVar'
[Hangup]
  == Registered application 'Hangup'
[Incomplete]
  == Registered application 'Incomplete'
[NoOp]
  == Registered application 'NoOp'
[Proceeding]
  == Registered application 'Proceeding'
[Progress]
  == Registered application 'Progress'
[RaiseException]
  == Registered application 'RaiseException'
[ResetCDR]
  == Registered application 'ResetCDR'
[Ringing]
  == Registered application 'Ringing'
[SayAlpha]
  == Registered application 'SayAlpha'
[SayDigits]
  == Registered application 'SayDigits'
[SayNumber]
  == Registered application 'SayNumber'
[SayPhonetic]
  == Registered application 'SayPhonetic'
[Set]
  == Registered application 'Set'
[MSet]
  == Registered application 'MSet'
[SetAMAFlags]
  == Registered application 'SetAMAFlags'
[Wait]
  == Registered application 'Wait'
[WaitExten]
  == Registered application 'WaitExten'
  == Manager registered action ShowDialPlan
  == Registered application 'Bridge'
-- Registered extension context 'parkedcalls' (0x198760) in table 0x1986c8; registrar: features
  == Registered application 'ParkedCall'
  == Registered application 'Park'
  == Manager registered action ParkedCalls
  == Manager registered action Park
  == Manager registered action Bridge
  == Manager registered action DBGet
  == Manager registered action DBPut
  == Manager registered action DBDel
  == Manager registered action DBDelTree
Asterisk Dynamic Loader Starting:
Asterisk Ready.

*CLI>

作者: 角色 時間: 2010-9-17 16:04

For 1.4.36 compilation, I have problems in find the command md5sum. Do you know where I can install it?

build_tools/make_buildopts_h: line 25: md5sum: command not found

YH

作者: 角色 時間: 2010-9-17 16:29

When I try to run "make menuselect", it gives the following errors:

[/opt/usr/src/asterisk-1.4.36] # make menuselect
CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts
make[1]: Entering directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
make[1]: `makeopts' is up to date.
make[1]: Leaving directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" cmenuselect
make[1]: Entering directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
make[1]: Nothing to be done for `cmenuselect'.
make[1]: Leaving directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" nmenuselect
make[1]: Entering directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
make[1]: Nothing to be done for `nmenuselect'.
make[1]: Leaving directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" gmenuselect
make[1]: Entering directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
make[1]: Nothing to be done for `gmenuselect'.
make[1]: Leaving directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
No menuselect user interface found. Install ncurses,
newt or GTK libraries to build one and re-rerun
'make menuselect'.

複製代碼

I have already installed ncurses and newt programms but I still failed to "make menuselect".

YH

作者: ckleea 時間: 2010-9-17 16:50

回復 23# Sam

It wsa somewhat I encountered before. I switch to Asterisk 1.4

作者: Sam 時間: 2010-9-18 00:28

For 1.4.36 compilation, I have problems in find the command md5sum. Do you know where I can install ...
角色發表於 2010-9-17 16:04

md5sum is in my system when I install the GNU coreutils.

作者: 角色 時間: 2010-9-18 00:33

Thank you for your information.
I am using QNAP TS-119. Let me try to install GNU coreutils.

YH

作者: Sam 時間: 2010-9-18 00:34

When I try to run "make menuselect", it gives the following errors:I have already installed ncurses ...
角色發表於 2010-9-17 16:29

Did "ldconfig -v" show libncurses ?

作者: 角色 時間: 2010-9-18 00:42

本帖最後由角色於 2010-9-18 07:14 編輯

Did "ldconfig -v" show libncurses ?
Sam 發表於 2010-9-18 00:34

Yes, it shows libncurses.so.5 -> libncurses.so.5.5.

YH

作者: Sam 時間: 2010-9-18 00:42

Something wrong with my Asterisk 1.6... don't know why it cannot locate all the config files although correct path already mentioned in asterisk.conf, I use "make DESTDIR=/opt install" but the binary still look for the default paths:

Could not load features.conf
No 'modules.conf' found, no modules will be loaded.
Unable to open Asterisk database '/var/lib/asterisk/astdb': No such file or directory
No 'modules.conf' found, no modules will be loaded.

作者: 角色 時間: 2010-9-18 00:44

md5sum is in my system when I install the GNU coreutils.
Sam 發表於 2010-9-18 00:28

After installing GNU coreutils, I re-make again. So far, it keeps on compiling without errors. Let us see what are the end results later.

YH

作者: Sam 時間: 2010-9-18 01:40

Oh, made a stupid mistake by forgot to uncomment [directories] in asterisk.conf,... My 1.6.2 in arm linux is up and running fine now!

作者: ckleea 時間: 2010-9-18 06:04

glad to hear you make it work.

作者: 角色 時間: 2010-9-18 15:02

I still have problem in compiling the Asterisk 1.4.36 source codes with the following errors:

[/opt/usr/src/asterisk-1.4.36] # make
CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts
make[1]: Entering directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
make[1]: `makeopts' is up to date.
make[1]: Leaving directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
In file included from chared.h:136,
from el.h:101,
from common.c:51,
from editline.c:4:
fcns.h:56:1: warning: "em_upper_case" redefined
fcns.h:50:1: warning: this is the location of the previous definition
In file included from editline.c:4:
common.c:73: error: expected identifier or '(' before numeric constant
common.c:302: error: expected identifier or '(' before numeric constant
common.c:353: error: expected identifier or '(' before numeric constant
common.c: In function 'ed_quoted_insert':
common.c:387: error: called object '8' is not a function
common.c: At top level:
common.c:397: error: expected identifier or '(' before numeric constant
common.c:441: error: expected identifier or '(' before numeric constant
common.c:466: error: expected identifier or '(' before numeric constant
common.c:485: error: expected identifier or '(' before numeric constant
common.c:498: error: expected identifier or '(' before numeric constant
common.c:524: error: expected identifier or '(' before numeric constant
common.c:537: error: expected identifier or '(' before numeric constant
In file included from editline.c:5:
emacs.c:122: error: expected identifier or '(' before numeric constant
emacs.c:288: error: expected identifier or '(' before numeric constant
emacs.c:368: error: expected identifier or '(' before numeric constant
emacs.c:382: error: expected identifier or '(' before numeric constant
emacs.c:416: error: expected identifier or '(' before numeric constant
emacs.c:470: error: expected identifier or '(' before numeric constant
emacs.c:483: error: expected identifier or '(' before numeric constant
In file included from editline.c:6:

複製代碼

Does anymore who have come across with it when compiling your codes under a QNAP TS-119 platform.

YH

作者: Sam 時間: 2010-9-18 23:17

For those who want to upgrade from Asterisk 1.4 to 1.6 please be very careful, 1.4 configuration will not work directly under 1.6 and many of the connect working well under 1.4 will be broken in 1.6, I am still having a hard time to flight with them....

作者: 角色 時間: 2010-9-19 06:22

In fact, under the Optware platform, there has Asterisk-1.6.2.12 ipkg install package. You do not need to compile it by yourself.

YH

作者: Sam 時間: 2010-9-19 10:20

You are right, I forgot to add the new Optware feeds to ipkg so I don't know 1.6 is already available, thanks!

作者: 角色 時間: 2010-9-19 10:26

In fact, I wanna compile the Asterisk tar ball by myself. However there are many errors coming out. I do not know how to fix it because I do not know the correct environment and libraries needed. As a result, I have to give up. I have no choice to switch of ipkg asterisk16 I want to.

YH

作者: Sam 時間: 2010-9-19 11:54

Maybe you can update to the latest C Dev Environment... I just figure out I can't use the latest Asterisk 1.6 ipk distribution because it is not compatible with my old glibc

作者: 角色 時間: 2010-9-19 12:17

Could you elaborate more about the way to update the latest C Development environment?

YH

作者: Sam 時間: 2010-9-19 23:18

Such as the C/C++ compiler, I've seen the above "error: expected identifier or '(' before numeric constant " usually reported on GCC 3.x.x while 4.1.1 should no longer complaint.

作者: 角色 時間: 2010-9-20 07:24

The one that I have has already been gcc version 4.2.3.

作者: ckleea 時間: 2010-10-22 04:35

The Asterisk Development Team is proud to announce the release of Asterisk
1.8.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4. For more information about
support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

The release of Asterisk 1.8.0 would not have been possible without the support
and contributions of the community. Since Asterisk 1.6.2, we've had over 500
reporters, more than 300 testers and greater than 200 developers contributed to
this release.

You can find a summary of the work involved with the 1.8.0 release in the
sumary:

http://svn.asterisk.org/svn/aste ... k-1.8.0-summary.txt

A short list of available features includes:

* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/aster ... CHANGES?view=markup

For a full list of changes in the current release candidate, please see the
ChangeLog:

http://downloads.asterisk.org/pu ... isk/ChangeLog-1.8.0

Thank you for your continued support of Asterisk!

作者: 角色 時間: 2010-10-22 07:17

Thanks CK for letting us know the most updated information. When I have time, I may set up one server for Asterisk 1.8 for evaluation.

YH

作者: ckleea 時間: 2010-10-22 08:59

YH,

When also have time, I will also do a compilation of asterisk 1.8. However, my problem right now is the slow processing speed for the VM image under my Windows XP.

作者: ckleea 時間: 2010-12-9 13:53

Asterisk 1.4.38, 1.6.2.15 & 1.8.1 Now Available

The release of Asterisk 1.4.38 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Add ability for Asterisk to try both the encoded and unencoded subscription
  URI for a match in hints.
  (Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)

* Set the caller id on CDRs when it is set on the parent channel.
  (Closes issue #17569. Reported, patched by tbelder)

* Ensure user portion of SIP URI matches dialplan when using encoded characters
  (Closes issue #17892. Reported by wdoekes. Patched by jpeeler)

* Fix a crash in res_jabber by ensuring that we don't alter memory after it's
  freed.
  (Closes issue #17387. Reported, tested by jmls. Patched by tilghman)

* Fix problem with qualify option packets for realtime peers never stopping.
  The option packets not only never stopped, but if a realtime peer was not in
  the peer list multiple options dialogs could accumulate over time.
  (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
  jpeeler)

* Multiple fixes related to Local channels.

The release of Asterisk 1.6.2.15 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* When using chan_skinny, don't crash when parking a non-bridged call.
  (Closes issue #17680. Reported, tested by jmhunter. Patched, tested by DEA)

* Add ability for Asterisk to try both the encoded and unencoded subscription
  URI for a match in hints.
  (Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)

* Set the caller id on CDRs when it is set on the parent channel.
  (Closes issue #17569. Reported, patched by tbelder)

* Ensure user portion of SIP URI matches dialplan when using encoded characters
  (Closes issue #17892. Reported by wdoekes. Patched by jpeeler)

* Resolve issue where Party A in an analog 3-way call would continue to hear
  ringback after party C answers.
  (Patched by rmudgett)

* Fix problem with qualify option packets for realtime peers never stopping.
  The option packets not only never stopped, but if a realtime peer was not in
  the peer list multiple options dialogs could accumulate over time.
  (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
  jpeeler)

* Multiple fixes related to Local channels.

The release of Asterisk 1.8.1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Fix issue when using directmedia. Asterisk needs to limit the codecs offered
  to just the ones that both sides recognize, otherwise they may end up sending
  audio that the other side doesn't understand.
  (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)

* Resolve issue where Party A in an analog 3-way call would continue to hear
  ringback after party C answers.
  (Patched by rmudgett)

* Fix playback failure when using IAX with the timerfd module.
  (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)

* Fix problem with qualify option packets for realtime peers never stopping.
  The option packets not only never stopped, but if a realtime peer was not in
  the peer list multiple options dialogs could accumulate over time.
  (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
  jpeeler)

* Fix issue where it is possible to crash Asterisk by feeding the curl engine
  invalid data.
  (Closes issue #18161. Reported by wdoekes. Patched by tilghman)

作者: ckleea 時間: 2011-1-15 07:12

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* 'sip notify clear-mwi' needs terminating CRLF.
(Closes issue #18275. Reported, patched by klaus3000)

* Patch for deadlock from ordering issue between channel/queue locks in
app_queue (set_queue_variables).
(Closes issue #18031. Reported by rain. Patched by bbryant)

* Fix cache of device state changes for multiple servers.
(Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested
by russellb)

* Resolve issue where channel redirect function (CLI or AMI) hangs up the call
instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos)

* Fix reloading of peer when a user is requested. Prevent peer reloading from
causing multiple MWI subscriptions to be created when using realtime.
(Closes issue #18342. Reported, patched by nivek.)

* Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0
so res_jabber doesn't think there is already an XMPP connection sending
device state. Also clean up CLI commands a bit.
(Closes issue #18272. Reported by klaus3000. Patched by Marquis42)

* Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
setting peer->cdr = NULL, set it to not post.
(Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)

* Fixes issue with outbound google voice calls not working. Thanks to az1234
and nevermind_quack for their input in helping debug the issue.
(Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... isk/ChangeLog-1.8.2

Thank you for your continued support of Asterisk!

The Asterisk Development Team has announced the release of Asterisk 1.6.2.16.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.16 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Fix cache of device state changes for multiple servers.
(Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested
by russellb)

* Resolve issue where channel redirect function (CLI or AMI) hangs up the call
instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos)

* Linux and *BSD disagree on the elements within the ucred structure. Detect
which one is in use on the system.
(Closes issue #18384. Reported, patched, tested by bjm, tilghman)

* app_followme: Don't create a Local channel if the target extension does not
exist.
(Closes issue #18126. Reported, patched by junky)

* Revert code that changed SSRC for DTMF.
(Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686.
Tested by cmbaker82)

* Resolve issue where REGISTER request with a Call-ID matching an existing
transaction is received it was possible that the REGISTER request would
overwrite the initreq of the private structure.
(Closes issue #18051. Reported by eeman. Patched, tested by twilson)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... /ChangeLog-1.6.2.16

Thank you for your continued support of Asterisk!

The Asterisk Development Team has announced the release of Asterisk 1.4.39. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.39 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Resolve issue where channel redirect function (CLI or AMI) hangs up the call
instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos)

* Fix bugs in saying numbers using the Swedish language syntax
(Closes issue #18355. Reported, patched by oej)

* Fix not stopping MOH when transfered local channel queue member is answered.
The problem here is only present when local channels are used with the MOH
passthru option as well as no optimization (/nm).
Patched by jpeeler.

* Improve handling of REGISTER requests with multiple contact headers.
Patched by jpeeler.

* app_followme: Don't create a Local channel if the target extension does not
exist.
(Closes issue #18126. Reported, patched by junky)

* Revert code that changed SSRC for DTMF.
(Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686.
Tested by cmbaker82)

* Resolve issue where REGISTER request with a Call-ID matching an existing
transaction is received it was possible that the REGISTER request would
overwrite the initreq of the private structure.
(Closes issue #18051. Reported by eeman. Patched, tested by twilson)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... sk/ChangeLog-1.4.39

Thank you for your continued support of Asterisk!

作者: bubblestar 時間: 2011-1-15 07:39

Thanks for the information. Hope this latest release can really resolve the bugs as they claim.

作者: ckleea 時間: 2011-1-15 08:28

回復 49# bubblestar

On the way to compile. Will see if any difference found

作者: 角色 時間: 2011-1-15 16:55

回復 50# ckleea

May I know the result?

YH

作者: ckleea 時間: 2011-1-15 16:57

So far so good. Those I want are there and working

作者: 角色 時間: 2011-1-15 16:59

It seems Asteriskf 1.8.2 becomes more and more popular. Have you installed the analogue card with the Asterisk 1.8.2 system?

YH

作者: ckleea 時間: 2011-1-15 17:01

Not. Only Bubblestar C-hing has this. I want to do so but may try the USB FXO instead.

作者: bubblestar 時間: 2011-1-15 17:12

Yes. Asterisk 1.8.2 has improved some of the bugs in IAX that I experienced before. Others are remaining stable so far.

作者: 角色 時間: 2011-1-15 17:34

Are all of you using YUM to upgrade or yourcompile the source code yourself?

YH

作者: ckleea 時間: 2011-1-15 17:38

WE compile from source code.
Mine is like this

svn co http://svn.asterisk.org/svn/asterisk/branches/1.8
cd 1.8
./configure
#this is ony for format MP3 - SVN required
contrib/scripts/get_mp3_source.sh
contrib/scripts/get_ilbc_source.sh
make menuselect
make

複製代碼

作者: bubblestar 時間: 2011-1-15 17:38

Seems YUM to upgrade to Asterisk 1.8.2 is not ready yet.

I compiled.

作者: ckleea 時間: 2011-2-23 07:33

The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.

These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an
issue that when decoding UDPTL packets, multiple stack and heap based arrays can
be made to overflow by specially crafted packets. Systems configured for
T.38 pass through or termination are vulnerable. The issue and resolution are
described in the AST-2011-002 security advisory.

For more information about the details of this vulnerability, please read the
security advisory AST-2011-002, which was released at the same time as this
announcement.

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... /ChangeLog-1.4.39.2
http://downloads.asterisk.org/pu ... /ChangeLog-1.6.1.22
http://downloads.asterisk.org/pu ... hangeLog-1.6.2.16.2
http://downloads.asterisk.org/pu ... s/ChangeLog-1.8.2.4

Security advisory AST-2011-002 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

Thank you for your continued support of Asterisk!

作者: bubblestar 時間: 2011-2-23 09:13

Upgraded last night. Thanks

作者: ckleea 時間: 2011-2-23 10:07

回復 60# bubblestar

Any problem encountered?

作者: bubblestar 時間: 2011-2-23 10:49

本帖最後由 bubblestar 於 2011-2-23 11:22 編輯

回復 61# ckleea

It was done at mid-night so only brief and random evaluation could be made. SIP, IAX, PSTN, Google Voice and IPTEL are fine. Cross Asterisk Server resources usage like using counterparts' SIP gateway to initiate call is also good as before.

作者: ckleea 時間: 2011-2-23 11:07

You are doing this really fast.

作者: ckleea 時間: 2011-3-1 05:40

Availability of 1
.8.3, 1.6.2.17 and 1.4.40

The Asterisk Development Team has announced the release of Asterisk 1.8.3. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.3 resolves several issues reported by the community
and would have not been possible without your participation. Thank you!

The following is a sample of the issues resolved in this release:

* Resolve duplicated data in the AstDB when using DIALGROUP()
(Closes issue #18091. Reported by bunny. Patched by tilghman)

* Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
(Closes issue #18464. Reported, patched by IgorG)

* Reworking parsing of mwi => lines to resolve a segfault. Also add a set of
unit tests for the function that does the parsing.
(Closes issue #18350. Reported by gbour. Patched by Marquis)

* When using cdr_pgsql the billsec field was not populated correctly on
unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman)

* Resolve memory leak in iCalendar and Exchange calendaring modules.
(Closes issue #18521. Reported, patched by pitel. Tested by cervajs)

* This version of Asterisk includes the new Compiler Flags option
BETTER_BACKTRACES which uses libbfd to search for better symbol information
within both the Asterisk binary, as well as loaded modules, to assist when
using inline backtraces to track down problems.
(Patched by tilghman)

* Resolve issue where no Music On Hold may be triggered when using
res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested
by francesco_r, rfrantik, one47)

* Resolve a memory leak when the Asterisk Manager Interface is disabled.
(Reported internally by kmorgan. Patched by russellb)

* Reimplemented fax session reservation to reverse the ABI breakage introduced
in r297486.
(Reported internally. Patched by mnicholson)

* Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)

* Resolve deadlock involving REFER.
(Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)

Additionally, this release has the changes related to security bulletin
AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... isk/ChangeLog-1.8.3

Thank you for your continued support of Asterisk!

[asterisk-users] Asterisk 1.6.2.17 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.6.2.17.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.17 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Resolve duplicated data in the AstDB when using DIALGROUP()
(Closes issue #18091. Reported by bunny. Patched by tilghman)

* Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev,
jthurman, elguero, zerohalo. Patched by tilghman)

* When using cdr_pgsql the billsec field was not populated correctly on
unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman)

* Resolve issue where re-transmissions of SUBSCRIBE could break presence.
(Closes issue #18075. Reported by mdu113. Patched by twilson)

* Fix regression causing forwarding voicemails to not work with file storage.
(Closes issue #18358. Reported by cabal95. Patched by jpeeler)

* This version of Asterisk includes the new Compiler Flags option
BETTER_BACKTRACES which uses libbfd to search for better symbol information
within both the Asterisk binary, as well as loaded modules, to assist when
using inline backtraces to track down problems.
(Patched by tilghman)

* Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo,
shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes.

* Resolve issue where no Music On Hold may be triggered when using
res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested
by francesco_r, rfrantik, one47)

* Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)

Additionally, this release has the changes related to security bulletin
AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... /ChangeLog-1.6.2.17

Thank you for your continued support of Asterisk!

The Asterisk Development Team has announced the release of Asterisk 1.4.40. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.40 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev,
jthurman, elguero, zerohalo. Patched by tilghman)

* Resolve issue where re-transmissions of SUBSCRIBE could break presence.
(Closes issue #18075. Reported by mdu113. Patched by twilson)

* Resolve issue in res_odbc where it may crash when a query fails.
(Closes issue #18243. Reported, patched by ks3)

* Fix CPU spike when pressing DTMF after agent login.
(Closes issue #18130. Reported by rgj. Patched by jpeeler)

* Fix cross-compiling issue.
(Closes issue #18301. Reported, patched by abelbeck)

* This version of Asterisk includes the new Compiler Flags option
BETTER_BACKTRACES which uses libbfd to search for better symbol information
within both the Asterisk binary, as well as loaded modules, to assist when
using inline backtraces to track down problems.
(Patched by tilghman)

* Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo,
shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes.

* Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)

Additionally, this release has the changes related to security bulletin
AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... sk/ChangeLog-1.4.40

Thank you for your continued support of Asterisk!

作者: bubblestar 時間: 2011-3-1 22:24

Thanks and upgraded again.

作者: 角色 時間: 2011-3-1 22:38

I am planning to set up my ATOM D525 + Analogue card three months later ..

YH

作者: ckleea 時間: 2011-3-2 06:12

回復 66# 角色
Which version of asterisk you plan to use?

作者: 角色 時間: 2011-3-2 07:13

回復 67# ckleea

Of course the most updated version.

YH

作者: ckleea 時間: 2011-3-2 08:18

回復 68# 角色

Are you planning to integrate skype as well? Siptosis is really a simple and good software to deal with.

作者: ckleea 時間: 2011-3-2 08:18

You may also consider to implement iaxmodem and avantfax. It is great for fax out service.

作者: bubblestar 時間: 2011-3-2 09:38

回復 70# ckleea

What is the requirement to integrate iaxmodem and avanfax into the system?

Appreciate if step-by-step installation can be shared with us.

Thanks

作者: ckleea 時間: 2011-3-2 10:39

Will do so later.

Both are software based. You need hylafax and apache installed
Installation of iaxmodem is fairly easy.
For avantfax, it is abit more complicated.

作者: bubblestar 時間: 2011-3-2 12:16

CDR BUG??

CDR in the latest build does not have proper record. It ALWAYS dispose the answer status as "NO ANSWER" even it was really answered. Also, the talk time begin and end were not counted, hence Billing time was ZERO.

However, in CLI, the call clearly shown ANSWERED.

Anyone encounters the same problem with the lastest Asterisk 1.8.3 ?

作者: ckleea 時間: 2011-3-4 13:19

The Asterisk Development Team is pleased to announce the release of
DAHDI-Linux and DAHDI-Tools version 2.4.1.

DAHDI-Linux 2.4.1, DAHDI-Tools 2.4.1, and DAHDI-Linux-Complete 2.4.1+2.4.1 are
available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

2.4.1 is a maintenance release of the DAHDI drivers and tools packages. Some of
the more notable changes are:

* Support for compilation against kernel versions from 2.6.9 up to and including
2.6.38-rc6.

* wct4xxp: PCI-express cards go through an extended reset at start by default.

* wcte12xp, wctdm24xxp: Disable read-line multiple PCI command, which increases
compatibility in some systems.

* xpp: Fixes init error for PRI devices with < 4 ports.

* tonezone: Add Macao, China to tone zone data.

* dahdi_genconf: Don't generate configurations that use channel 16 on E1 CAS.

For a full list of changes in these releases, please see the ChangeLogs at
http://svn.asterisk.org/svn/dahdi/linux/tags/2.4.1/ChangeLog and
http://svn.asterisk.org/svn/dahdi/tools/tags/2.4.1/ChangeLog

Issues found in these release candidates can be reported in the DAHDI-linux or
DAHDI-tools project at https://issues.asterisk.org

Thank you for your continued support of Asterisk!

作者: bubblestar 時間: 2011-3-19 23:34

Asterisk 1.8.3.2 is released on March 17, 2011

http://www.asterisk.org/downloads

作者: ckleea 時間: 2011-3-31 12:24

回復 74# ckleea

You need this version of Dahdi before you can enjoy USB 3G modem for phone call with asterisk.

作者: ckleea 時間: 2011-4-12 06:58

The Asterisk Development Team announces the release of DAHDI-Linux 2.4.1.2.

DAHDI-Linux 2.4.1.2 and DAHDI-Linux-Complete 2.4.1.2+2.4.1 are
available for immediate download at:

http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

2.4.1.2 is a maintenance release that resolves a conflict with RHEL 5.6. RHEL
5.6 backported the definition of dev_name from kernel 2.6.26. DAHDI also had
this definition backported. The result was that DAHDI would fail to compile.
The issue was originally reported in [1].

[1] https://issues.asterisk.org/view.php?id=18992

Issues found in these releases can be reported in the DAHDI-linux project at
https://issues.asterisk.org

Thank you for your continued support of Asterisk!

作者: 角色 時間: 2011-4-12 07:19

Thank for CK info.

YH

作者: ckleea 時間: 2011-4-12 07:27

Only need if you upgrade centos to 5.6 But be aware that you may encounter problem in apache.

作者: ckleea 時間: 2011-4-22 09:54

Secruity Update

The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3.

These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two
issues:

* File Descriptor Resource Exhaustion (AST-2011-005)
* Asterisk Manager User Shell Access (AST-2011-006)

The issues and resolutions are described in the AST-2011-005 and AST-2011-006
security advisories.

For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-005 and AST-2011-006, which were released at the
same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLog:

http://downloads.asterisk.org/pu ... /ChangeLog-1.4.40.1
http://downloads.asterisk.org/pu ... /ChangeLog-1.6.1.25
http://downloads.asterisk.org/pu ... hangeLog-1.6.2.17.3
http://downloads.asterisk.org/pu ... s/ChangeLog-1.8.3.3

Security advisory AST-2011-005 and AST-2011-006 are available at:

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

Thank you for your continued support of Asterisk!

作者: bubblestar 時間: 2011-5-14 12:28

Asterisk 1.8.4 has been released.

I'm on my way in compilation.

作者: ckleea 時間: 2011-5-14 12:41

回復 81# bubblestar

I notice this about 2 days but do not have time to upload here.

Other important news for us is
Microsoft acquired skype. I think it will bring serious impact to many many users. Given the quality of microsoft product, I am afriad the future of skype would contain too many unnecessary functions. Just too big a file.

作者: ckleea 時間: 2011-5-14 12:43

For the interest of our members

Asterisk 1.8.4 Now Available.

Asterisk_OSR_ The Asterisk Development Team has announced the release of Asterisk 1.8.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.4 resolves several issues reported by the community.
Without your help this release would not have been possible. Thank you!

Below is a sample of the issues resolved in this release:

* Use SSLv23_client_method instead of old SSLv2 only.
(Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell
and chazzam.

* Resolve crash in ast_mutex_init()
(Patched by twilson)

* Resolution of several DTMF based attended transfer issues.
(Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo,
shihchuan, grecco. Patched by rmudgett)

NOTE: Be sure to read the ChangeLog for more information about these changes.

* Resolve deadlocks related to device states in chan_sip
(Closes issue #18310. Reported, patched by one47. Patched by jpeeler)

* Resolve an issue with the Asterisk manager interface leaking memory when
disabled.
(Reported internally by kmorgan. Patched by russellb)

* Support greetingsfolder as documented in voicemail.conf.sample.
(Closes issue #17870. Reported by edhorton. Patched by seanbright)

* Fix channel redirect out of MeetMe() and other issues with channel softhangup
(Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb.
Patched by russellb)

* Fix voicemail sequencing for file based storage.
(Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by
jpeeler)

* Set hangup cause in local_hangup so the proper return code of 486 instead of
503 when using Local channels when the far sides returns a busy. Also affects
CCSS in Asterisk 1.8+.
(Patched by twilson)

* Fix issues with verbose messages not being output to the console.
(Closes issue #18580. Reported by pabelanger. Patched by qwell)

* Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by
alecdavid, Irontec, ZX81, cmaj)

Includes changes per AST-2011-005 and AST-2011-006 For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pu ... isk/ChangeLog-1.8.4

Information about the security releases are available at:

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

Thank you for your continued support of Asterisk!

作者: bubblestar 時間: 2011-5-15 17:12

本帖最後由 bubblestar 於 2011-5-15 17:27 編輯

Anybody can update to Asterisk 1.84 successfully?? I failed. Googling and find that these problem has been there in 1.8.4rc1, rc2, rc3. Don't know whether it is really fixed or not.

Before update, I stop the current process of Asterisk. After update and when entering to CLI as usual by typing

> asterisk -grv

I got Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?).

> asterisk -cvvv to see whether it can give me some hints

> ps -A | grep asterisk and ps aux | grep asterisk to kill pid and retry to reboot but still no luck.

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作者: ckleea 時間: 2011-5-16 17:52

回復 84# bubblestar

try to stop all asterisk process

by

service asterisk stop
chkconfig asterisk off

reboot

redownload asterisk 1.84
recompile
start asterisk again

if ok,

then chkconfig asterisk on to allow auto-start

Something worry with the source code or the autostart script i.e. /etc/init.d/asterisk

作者: ckleea 時間: 2011-5-16 21:15

Try rename chan_datacard.so to other. May be some incompatibilty

作者: bubblestar 時間: 2011-5-16 22:21

回復 86# ckleea

Not today but will try using with your suggested method later. In my last 10 or 20 + updates, I followed the same procedure without problem. Hope the suspect and culprit is chan_datacard.so as you said because this was the major feature that I added since late update.

作者: bubblestar 時間: 2011-5-17 15:36

本帖最後由 bubblestar 於 2011-5-17 15:44 編輯

回復 84# bubblestar

[Problem Solved]

After one more trial (in fact, I had tried to install 7-8 times totally in these 2 days), I can manage to update my Asterisk from version 1.8.3.3 to 1.8.4 finally.

The culprit that causes the problem in entering CLI after update before, as ckleea c-hing said, is "datacard".  I suggest those who plays with or has installed chan_datacard to disable the following configuration file before update to Asterisk 1.8.4.  Otherwise, you will encounter entering issue into CLI.

Disable datacard.conf under extensions.conf is quite enough.  We don't need to disable datacard_extensions.conf and chan_datacard.so respectively.  For more comprehesive and safety sake, I disable all the 3 of them.

Then update Asterisk 1.8.4 as usual and the problem to access CLI will be resolved automatically.  After that, I tried to re-enable those 3 files but unfortunately, the problem comes back again.  Apparently, there is some establishment conflict between datacard and the latest Asterisk 1.8.4.  Only "Segmentation Fault" displays on the screen.  Hope the next version will fix the problem.

It is clear that , for the time being, the features of SMS and Telephone call over 3G USB stick via Asterisk has to be suspended.  So frustrating and disappointing hit to me.

Would ckleea c-hing, if have time, please tell us whether your new chan_dongle can get along with Asterisk 1.8.4.

Many thanks

作者: ckleea 時間: 2011-5-17 15:57

回復 88# bubblestar

I will tell you tonight because I have not tested the dial in and out functions. Also voice quality. So this is why I do not put up the reference.

Try to recompile the chan_datacard source again with the new asterisk 1.8.4 sources. It may be that the chan_datacard.so contains old code that make it not working.

作者: ckleea 時間: 2011-5-17 16:01

回復 88# bubblestar

Most simple is to rename the file chan_datacard.so to chan_datacard.so.old. It is located in /usr/lib/asterisk/modules

作者: bubblestar 時間: 2011-5-17 16:13

本帖最後由 bubblestar 於 2011-5-17 16:29 編輯

回復 89# ckleea

Great minds think alike.

Your suggestion also just comes to my mind that I should re-compile chan_datacard. The idea is originated from DAHDI. I recall that whenever there are any new revisions in CentOS or Kernel updates, I have to re-compile dahdi once again or it cannot be recognized. Will try it tonight after making backup.

作者: ckleea 時間: 2011-5-17 16:19

互相合作最重要。

作者: bubblestar 時間: 2011-5-17 16:24

本帖最後由 bubblestar 於 2011-5-17 16:28 編輯

回復 90# ckleea

Very strange, just rename chan_datacard.so in /usr/lib/asterisk/modules has no effect and in vain.

What I am doing now to make it work is to:

1.  add a line "noload chan_datacard.so" in modules.conf under /etc/asterisk directory.
2.  (This is an important step) rename datacard.conf to datacard.disabled.conf under /etc/asterisk directory.
3.  chan_datacard.so in /usr/lib/asterisk/modules remained intact.

作者: ckleea 時間: 2011-5-17 16:57

strange. change the modules name to an extension other than so will do.

作者: bubblestar 時間: 2011-5-17 17:04

Haha, agree with what you said. So that why you describe it as "Strange" and I describe it as "Very strange".

作者: bubblestar 時間: 2011-5-17 19:42

回復 91# bubblestar

ckleea c-hing,

Glad to advise that our judgement is correct. After re-compiling chan_datacard, it can now be recognized by latest Asterisk 1.8.4 without the need to disable any modules and config files .

I think that this also becomes our upgrading routine whenever new Asterisk release or CentOS, or Kernal update is made.

作者: ckleea 時間: 2011-5-17 19:44

Not always. But some modules need recompiling.
E.g. connector to TTS e.g. espeak, etc

作者: ckleea 時間: 2011-5-17 20:21

回復 96# bubblestar

Bubblestar,

What kind of packages you are downloading from asterisk?

Trunk or branches?

You may try my scripts

#get branch
svn co http://svn.asterisk.org/svn/asterisk/branches/1.8 asterisk-1.8
#get trunk
svn checkout http://svn.asterisk.org/svn/asterisk/trunk asterisk-1.8
cd asterisk-1.8
./configure
#this is ony for format MP3 - SVN required
contrib/scripts/get_mp3_source.sh
contrib/scripts/get_ilbc_source.sh
make menuselect
make

作者: ckleea 時間: 2011-5-17 21:15

My update here: work as expected. No problem here.

May be you have not yet gained much experience in dealing with linux server.

do post your help here and see I can help.

作者: ckleea 時間: 2011-5-17 21:18

Forgot to mention. My version number is

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