Asterisk-Addons 1.6.1.0 Now Available

The Asterisk Development Team is pleased to announce the release of Asterisk-Addons 1.6.1.0. Asterisk-Addons 1.6.1.0 is available for immediate download at http://downloads.digium.com/pub/asterisk/

This release maintains compatibilty of the Asterisk-Addons package with the Asterisk 1.6.1 series of releases, since it is possible for API changes to occur between 1.6.x releases. Additionally, several minor issues have been resolved.

For a summary of the changes in this release, please see the release summary:

http://svn.digium.com/svn/asterisk-addons/tags/1.6.1.0/asterisk-addons-1.6.1.0-summary.txt

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

http://svn.digium.com/svn/asterisk-addons/tags/1.6.1.0/ChangeLog

For those upgrading from previous versions, it is advisable to review the UPGRADE.txt file:

http://svn.digium.com/svn/asterisk-addons/tags/1.6.1.0/UPGRADE.txt

Thank you for your continued support of Asterisk!

Asterisk 1.6.1.0 Now Available

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.1.0. Asterisk 1.6.1.0 is available for immediate download at http://downloads.digium.com/pub/asterisk/

This is the first release in the 1.6.1 branch, which has additional features added since 1.6.0. Please see the CHANGES file for more information about the additional functionality

For those upgrading from previous versions of Asterisk, it is advisable to review the UPGRADE.txt file:

http://svn.digium.com/svn/asterisk/tags/1.6.1.0/UPGRADE.txt

Some highlights about changes in this release:

----------------------------------------------

* It is now possible to specify a pattern match as a hint. Once a phone subscribes to something that matches the pattern a hint will be created using the contents and variables evaluated.

* IAX2 encryption support has been improved to support periodic key rotation within a call for enhanced security.  The option "keyrotate" has been provided to disable this functionality to preserve backwards compatibility with older versions of IAX2 that do not support key rotation.

* res_odbc no longer has a limit of 1023 total possible unshared connections, as some people were running into this limit.  This limit has been increased to 4.2 billion.

* Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given adaptive capabilities.  What this means in practical terms is that if your realtime table lacks critical fields, Asterisk will now emit warnings to that effect.  Also, some of the realtime drivers have the ability (if configured) to automatically add those columns to the table with the correct type and length.

* Config file variables may now be appended to, by using the '+=' append operator.  This is most helpful when working with long SQL queries in func_odbc.conf, as the queries no longer need to be specified on a single line.

* Many many other changes that are too numerous to list here. See:

http://svn.digium.com/svn/asterisk/tags/1.6.1.0/CHANGES

For a summary of the changes in this release, please see the release summary:

http://svn.digium.com/svn/asterisk/tags/1.6.1.0/asterisk-1.6.1.0-summary.txt

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

http://svn.digium.com/svn/asterisk/tags/1.6.1.0/ChangeLog

The following list of issues were resolved with the participation of the community, and this release would not have been possible without your help!

* Allow disconnect feature before a call is bridged

- Closes issue #11583. Submitted by sobomax. Tested and additional coding by sobomax, dvossel, murf.
* Update app_fax to work with spandsp-0.0.6

- Closes issue #13688. Reported by and patched by irroot.
* chan_h323 with H323Plus for TRUNK (SVN rev. 89183)

- Closes issue #11261. Reported by vhatz. Patched by jthurman.

* Wrong usage of sscanf with use of uninitialized variable caused accidental parsing of RTP/SAVP

- Closes issue #14000. Reported and patched by folke.
*  Realtime peers are never qualified after 'sip reload'

- Closes issue #14196. Reported, tested, and patched by pdf.

Thank you for your continued support of Asterisk!

Asterisk 1.6.1.0-rc5 Now Available

The Asterisk Development Team is pleased to announce the fifth release candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc5 is available for immediate download at http://downloads.digium.com/pub/asterisk/

This release fixes a couple of issues with realtime music on hold that could cause Asterisk to crash, and an issue that caused hungup channels to stay up, leading to 100% CPU usage. Additionally, several minor issues and edge case scenarios have been resolved.

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

http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc5/ChangeLog

Issues found in this release candidate can be reported at http://bugs.digium.com

Thank you for your continued support of Asterisk!

Randomizing your voicemail greetings with Asterisk to annoy friends and family!

Do you run Asterisk for your house as well as for your business? Have you ever wanted to play a random voicemail greeting to the house callers? Just so they aren't always hearing the same thing every time they call? Well, with this little trick you can do it very easily.

First, find yourself some Funny Answering Machine messages or other such thing that you want to rotate around. These will usually come in .wav or .mp3 format, which you'll have to convert, using information from voip-info on converting from wav to gsm for use with asterisk.

After you've converted your files, you can follow these simple steps to get some random voicemail.

First, make a directory for your random voicemail gsm files

# mkdir /var/spool/asterisk/random-vm

Now, copy all your gsm files to this directory

# mv /home/admin/*.gsm /var/spool/asterisk/random-vm

Now, delete your old unavailable messages from your voicemail directory. In this case it's extension 7000, yours might differ.

# cd /var/spool/asterisk/voicemail/device/7000/
# rm unavail.gsm unavail.wav unavail.WAV busy.gsm busy.wav busy.WAV

Now, create a symlink from busy.gsm to unavail.gsm

# ln -s unavail.gsm busy.gsm

Now you just need to setup a little script to randomize the voicemail greetings.

# cd /var/spool/asterisk/random-vm
# nano vmscript.sh

And using your favourite editor (nano for us.. yes, we're not that nerdy) paste the following in.

#!/bin/sh

# number of voicemails to rotate is 6 (0-5). Change to match yours.
number=`expr $(date +%S) % 6`

# remove the old voicemail message
rm /var/spool/asterisk/voicemail/device/7000/unavail.gsm

# link the unavailable message to one of the random ones, yo!
ln -s /var/spool/asterisk/random-vm/unavail-$number.gsm /var/spool/asterisk/voicemail/device/7000/unavail.gsm

# done!

Now, exit and save the file. Remember to change the paths, and amount of voicemail greetings as required.

All you should have to do now, is change the permissions, and then add it to cron and you'll be set.

Change permissions like this

# chown -R asterisk:asterisk /var/spool/asterisk/random-vm
# chmod +x /var/spool/asterisk/random-vm/vmscript.sh

Add it to cron like this

# crontab -e

then add the following line

# to rotate voicemail messages
*/1 * * * * /var/spool/asterisk/random-vm/vmscript.sh

Exit and save your crontab. Which will rotate them every minute. Obviously, if you get like 30 calls a minute you may run into trouble, but for a simple house line, this works great, and always keeps callers guessing!

UPDATE

turns out it wasn't fully randomizing the messages as I intended. To get this to work, simply change the script above from

#!/bin/sh

to

#!/bin/bash

and then change

number=`expr $(date +%S) % 6`

to

number=$(($RANDOM % 6))

Now it'll rotate as expected.

UPDATE

Asterisk 1.6.0.6 Now Available!

The Asterisk.org development team is proud to announce the release of Asterisk 1.6.0.6. This release is available for download from http://downloads.digium.com/.

This release is a significant bug fix update for the 1.6.0 release series.

In addition, this release is recommended for all users of the Asterisk GUI.

Two issues with the manager interface have been resolved. The first being with the manager interface improperly handling async. requests from the GUI (see issue #14364). It resulted in manager session file descriptors being improperly destroyed and overwritten. The other being an issue with the Originate action that would cause issues with the GUI. They have both been resolved in this release.

The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help!

* Set the initiator attribute to lowercase in our replies when receiving calls.

This attribute contains a JID that identifies the initiator of the GoogleTalk

voice session. The GoogleTalk client discards Asterisk's replies if the

initiator attribute contains uppercase characters.

- Closes issue #13984. Submitted and tested by: jcovert

* Fix odd "thank you" sound playing behavior in app_queue.c

- Closes issue #14227. Reported and tested by: caspy

* Don't do an SRV lookup if a port is specified RFC 3263 says to do A record

lookups on a hostname if a port has been specified, so that's what we're

going to do. See section 4.2.

- Closes issue #14419. Reported and patched by: klaus3000

* Properly set "seen" and "unseen" flags when moving messages from the new to

the old folder when using IMAP for voicemail storage.

- Closes issue #13905. Reported and patched by: jaroth

* Fix devicestate problems for "always-on" agent channels.

- Closes issue #14173. Reported by: nathan. Tested by: nathan, aramirez

For a full list of changes, see the ChangeLog:

http://svn.digium.com/svn-view/asterisk/tags/1.6.0.6/ChangeLog?view=markup

Thank you for your support of Asterisk!

dahdi-linux 2.1.0.4 released

The Asterisk development team has released dahdi-linux 2.1.0.4 This release is available for immediate download from http://downloads.digium.com/pub/telephony/dahdi-linux.

This release fixes a regression from dahdi-linux 2.1.0 in which it was possible for the kernel to panic when conferencing channels together.

Please see http://bugs.digium.com/view.php?id=14183 for more information.

The complete change log can be read at:

http://downloads.digium.com/pub/telephony/dahdi-linux/releases/ChangeLog-2.1.0.4

Thanks for your continued support of Asterisk

Asterisk 1.6.1 Release Candidate 1 Now Available

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.6.1, tagged as version 1.6.1-rc1. Release candidate 1.6.1-rc1 is available for immediate download at http://downloads.digium.com/.

This release candidate includes a fix to SIP registrations when using realtime.

Additional crash issues have been resolved, in addition to making chan_sip more robust in handling unique scenarios. Issues found in this release candidate can be reported at http://bugs.digium.com/.

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

ChangeLog:

http://svn.digium.com/view/asterisk/tags/1.6.1-rc1/ChangeLog?view=co

Also see the CHANGES file for useful information about what is new in Asterisk 1.6.1. See the CHANGES file at:

http://svn.digium.com/view/asterisk/tags/1.6.1-rc1/CHANGES?view=co

Thank you for your continued support of Asterisk!

Next Page »