This is a minor release with a single fix for chan_ooh323c from issue #14116, which could cause a crash within the channel driver. If you are not using chan_ooh323c, there is no need to upgrade.

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

http://svn.digium.com/svn/asterisk-addons/tags/1.4.8/ChangeLog

Thank you for your continued support of Asterisk!

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

This update for Asterisk includes a security fix for chan_iax2. Please see the associated security adivisory for more details:

http://downloads.digium.com/pub/security/AST-2009-001.html

These updates are a fix to a previous security release (released as versions 1.2.31, 1.4.22.1, and 1.6.0.3).

The new versions are being released after additional testing revealed some issues with the way that scanning for users was blocked. Those issues have been corrected in this release.

This security issue affects the 1.2, 1.4, and 1.6 series of Asterisk.

Also note, that Asterisk 1.6.0.4-rc1 was released yesterday prior to the security update. That release has been removed as there will be no 1.6.0.4 release, but rather will be reincarnated as 1.6.0.6-rc1. The reason for the dead release is to avoid 5 digit release numbers.

ChangeLogs for the various releases are available at:

http://downloads.digium.com/pub/asterisk/ChangeLog-1.2.31.1

http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.22.2

http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.23.1

http://downloads.digium.com/pub/asterisk/ChangeLog-1.6.0.5

Thank you for your continued support of Asterisk!

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

* Fixed datastore handling crashes in app_queue.c.

- Closes issues 14086. Reported by ZX81. Tested by: ZX81, festr.

* Update autosupport script to supply info for both Zaptel and DAHDI

in 1.4 and be sure to run dahdi_test in 1.6.x and trunk instead of

zttest.

- Closes issue 14132. Reported by and patch contributed by: dsedivec

* Don't overflow when paging more than 128 extensions. The number of available

slots for calls in app_page was hardcoded to 128. Proper bounds checking

was not in place to enforce this limit, so if more than 128 extensions were

passed to the Page() app, Asterisk would crash. This patch instead

dynamically allocates memory for the ast_dial structures and removes the

(non-functional) arbitrary limit. This issue would have special importance

to anyone who is dynamically creating the argument passed to the Page

application and allowing more than 128 extensions to be added by an outside

user via some external interface.

- Closes issue 14217. Reported by and patch contributed by: a_villacis

* Do not crash if we are not passed a followme ID.

- Closes issue 14106. Reported by and patch contributed by: ys

* Change the way the T.38 SDP attribute handling to make it more liberal in

what it accepts, and more strict in what it sends.

- Closes issue 13967. Reported by: linulin. Patches contributed by:

arcivanov

In addition to the issues cited above, several issues related to call parking have been resolved, thereby making Asterisk call parking more robust.

See issues 13820, 13747, 14066, 14228, 13854, 12854, and 13139 for more information. Special thanks to otherwiseguy, davidw, bluefox, waverly360, kobaz, Adam Lee, krisk84, and murf.

For a full list of changes, see the ChangeLog:

http://svn.digium.com/svn-view/asterisk/tags/1.4.23/ChangeLog?view=markup

Thank you for your support of Asterisk!

This release candidate is available for download from http://downloads.digium.com/.

A number of critical issues have been resolved since the last release candidate for 1.4.23.  We hope to have this be the final release candidate.  If all goes well, 1.4.23 will be released early next week.

For a full list of changes, see the ChangeLog:

http://svn.digium.com/svn-view/asterisk/tags/1.4.23-rc4/ChangeLog?view=markup

Testing of this release candidate would be very much appreciated.  One area that has received a lot of changes is the handling of call parking, so that is an area that could use some special attention in testing.

Please report any issues found in testing to http://bugs.digium.com/.

Thank you for your support of Asterisk!

With this software they already had the Languages module available to them, so they were able to create one main IVR, and then several branching IVR's dependant on the language that was selected.

In this case, English and French. So, that's all well and good, then we noticed that when you dial the company directory from the French menu, the prompt for the instructions (and only that prompt) is always played in english, instead of our desired language of french. This is really weird. So, we debugged it a little bit and found this in the logs:

-- Executing [3@app-languages:2] Set("DAHDI/1-1", "CHANNEL(language)=fr") in new stack
-- <DAHDI/1-1> Playing 'dir-intro-oper' (language 'fr')
directory|default|from-did-direct|lo: -- Playing 'dir-instr' (language 'en')

Hmm, that's weird - it says it's playing it in french, but it's playing it in english that first time. Then it goes on to correctly say that it's playing the english soundfile, not the french one. Hm, even weirder. So, maybe it's permissions.

We checked the permissions in /var/lib/asterisk/sounds and /var/lib/asterisk/sounds/fr and everything was correct. They were owned by the asterisk user and asterisk group, we don't run as root and neither should you. The permissions were rw-r-r so that was fine too. Hmmmf. So, now we began to search google. We found one post on the subject from the freepbx bug tracker where the user was basically told it was his problem, so, we thought it was our problem too and continued searching for another hour or two trying everything possible:

• Replaced the sound files a couple times, deleting in between
• Re-did the permissions
• Symlinked them to various directories
• Added a hard coding of "fr/" to all soundfiles in the FreePBX AGI - Directory
• Ensured we have support for en and fr enabled
• Manually created the Directory with another Language and Misc Destination in FreePBX (complex!)
• Asked on IRC

Well, if you're reading this, none of that probably worked for you either. So, this is what we did - not sure how "right" it is, but it avoids this issue if you're trying to run your IVR with a company directory in english and french. It's pretty simple too.

First, login to your FreePBX interface.

Then click over to the Tools tab on the left.

Go to Custom Destinations, and Create a new one

Custom Destination: from-internal-custom,s,1
Name: French-Directory

Now press the submit button.

Now Flip over to the Setup Tab of FreePBX and go to your French IVR menu with the Directory in it.

Change the destination to go from the default (#) or whatever you have it set to, to instead go to your new Custom Destination (French-Directory)

Now you're done the setup in FreePBX.

SSH to your PBX system, and then edit the custom extensions file

# sudo nano /etc/asterisk/extensions_custom.conf

Now Paste the following in there

[from-internal-custom]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(LANGUAGE()=fr)
;exten => s,n,AGI(directory,${DIR-CONTEXT},from-did-direct,${DIRECTORY:0:1}${DIRECTORY_OPTS}o)
exten => s,n,Directory(default|from-did-direct|lo)
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup()
exten => o,1,Goto(from-internal,${OPERATOR_XTN},1)

Exit and save the file then reload asterisk.

# sudo asterisk -rx "reload"

Now, you should test the calling into your menu to make sure the proper directory is working. Does for us. Comments are open if anyone has better suggestions, or a way to fix the directory agi that comes with FreePBX to properly support the instructions/intro being played in french (or any other language) instead of only english.

This update for Asterisk includes a security fix for chan_iax2. Please see the associated security adivisory for more details:

http://downloads.digium.com/pub/security/AST-2009-001.html

This security issue affects the 1.2, 1.4, and 1.6 series of Asterisk. Asterisk releases 1.2.31, and 1.4.22.1 only contain the security fix.

Asterisk release 1.6.0.3 has additional changes from 1.6.0.2, including a fix to the Makefile that caused menuselect to break in certain cases.

Also, some issues related to memory leaks and more appropriate SIP dialog responses when receiving 4XX messages have been fixed.

ChangeLogs for the various releases are available at:

http://downloads.digium.com/pub/asterisk/ChangeLog-1.2.31

http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.22.1

http://downloads.digium.com/pub/asterisk/ChangeLog-1.6.0.3

Thank you for your continued support of Asterisk!

This release candidate contains multiple fixes since 1.4.23-rc2 including issues with transfers and DTMF. For a full list of the changes in this release, please see the ChangeLog:

http://svn.digium.com/view/asterisk/tags/1.4.23-rc3/ChangeLog?view=markup

Thank you for your continued support of Asterisk!

December 5th, 1999 was the initial release of Asterisk by Mark Spencer. We'll be celebrating this by gathering as usual at 12 Noon Eastern (9AM Pacific, 10 MST, 11 Central, 5PM UK and Western EU) for the VoIP Users Conference.

You can get all the dial in information at http://VoipUsersConference.org including info on a SipAddHeader() kludge to avoid DTMF problems.

IRC is Freenode.net #voip-users-conference join this even if you can't call in.

Call via SIP: talkshoe@vuc.onsip.com (thanks to OnSip.com) Call via PSTN (724) 444-7444 DTMF 22622# 1#

or try this: 7463#22622#1@proxy.ideasip.com (thanks to IdeaSIP.com)

or to just look up talkshoe server IP: ts.x2z.eu (thanks top me for the DNS record)

We start about 15 minutes to the hour with an informal chat.

Join us anytime, but especially, grab a virtual beer and join us Friday the 5th.

/r

This update for Asterisk includes a fix for a regression introduced in Asterisk 1.2.30 and Asterisk 1.4.21.2 and has existed in the Asterisk 1.6 branch since release. All releases with the exception of Asterisk 1.2.30.3 also contain a vast assortment of bugfixes in these releases. For a full list of changes, see the ChangeLogs:

http://svn.digium.com/view/asterisk/tags/1.2.30.3/ChangeLog?view=markup
http://svn.digium.com/view/asterisk/tags/1.4.23-rc2/ChangeLog?view=markup
http://svn.digium.com/view/asterisk/tags/1.6.0.2/ChangeLog?view=markup
http://svn.digium.com/view/asterisk/tags/1.6.1-beta3/ChangeLog?view=markup
http://svn.digium.com/view/asterisk-addons/tags/1.6.0.1/ChangeLog?view=markup
http://svn.digium.com/view/asterisk-addons/tags/1.6.1-rc2/ChangeLog?view=markup

Thank you for your continued support of Asterisk!