Asterisk-Addons 1.6.1.0 Now Available : Voip Phreak - asterisk

By: Voip Phreak  09-12-2011
Keywords: Asterisk

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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.

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

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

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

Thank you for your continued support of 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:

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:

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

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

Thank you for your continued support of 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:

Thank you for your continued support of Asterisk!

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

# ln -s unavail.gsm busy.gsm

# 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

# done!

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

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

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

- 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:

Thank you for your support of Asterisk!

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.

The complete change log can be read at:

Thanks for your continued support of Asterisk

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

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

ChangeLog:

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

Thank you for your continued support of Asterisk!

The information in this article was current at 06 Dec 2011

Keywords: Asterisk

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