ckleea

回復 74# ckleea

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

ckleea

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!

ckleea

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

ckleea

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!

ckleea

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.

ckleea

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!

ckleea

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

ckleea

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

ckleea

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.

ckleea

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

ckleea

互相合作最重要。

ckleea

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

ckleea

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

ckleea

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

ckleea

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.