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 <title>Computer Tech Union - Asterisk Programming</title>
 <link>http://www.ctunion.com/taxonomy/term/30/0</link>
 <description>Post reguarding Asterisk Programming</description>
 <language>en</language>
<item>
 <title>converting mp3's for use as polycom ringers</title>
 <link>http://www.ctunion.com/node/620</link>
 <description>&lt;p&gt;first use mhWaveEdit and convert to 8 bit 8000hz mono&lt;/p&gt;
&lt;p&gt;then&lt;/p&gt;
&lt;p&gt;sox phonesring.wav -U -c1 phonesring1.wav&lt;/p&gt;
&lt;p&gt;to put in g711&lt;/p&gt;
&lt;p&gt;# file phonesring.wav&lt;br /&gt;
phonesring.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 8 bit, mono 8000 Hz&lt;/p&gt;
</description>
 <pubDate>Tue, 05 Feb 2008 15:47:13 -0800</pubDate>
</item>
<item>
 <title>Queues with callback members Tutorial</title>
 <link>http://www.ctunion.com/node/241</link>
 <description>&lt;p&gt;Setting up Call Queues -- A Tutorial&lt;/p&gt;
&lt;p&gt;(This was pulled directly from the 1.4 asterisk docs folder included with the source)&lt;/p&gt;
&lt;p&gt;Pardon, but the dialplan in this tutorial will be expressed&lt;br /&gt;
in AEL, the new Asterisk Extension Language. If you are&lt;br /&gt;
not used to its syntax, we hope you will find it to some&lt;br /&gt;
degree intuitive. If not, there are documents explaining&lt;br /&gt;
its syntax and constructs.&lt;/p&gt;
&lt;p&gt;======  Configuring Call Queues&lt;/p&gt;
&lt;p&gt;First of all, set up call queues in queue.conf&lt;/p&gt;
&lt;p&gt;Here is an example:&lt;/p&gt;
&lt;p&gt;   =========== queues.conf ===========&lt;br /&gt;
   | ; Cool Digium Queues            |&lt;br /&gt;
   | [general]                       |&lt;/p&gt;
</description>
 <pubDate>Tue, 31 Jul 2007 08:07:38 -0700</pubDate>
</item>
<item>
 <title>convert wav files to gsm for use with asterisk / convert many mp3 to gsm</title>
 <link>http://www.ctunion.com/node/191</link>
 <description>&lt;p&gt;I have to use this all the time to convert my wav files to gsm for use in asterisk.&lt;/p&gt;
&lt;p&gt;sox foo.wav -r 8000 -c1 foo.gsm resample -ql&lt;/p&gt;
&lt;p&gt;make sure you have sox installed&lt;/p&gt;
&lt;p&gt;# apt-get install sox&lt;/p&gt;
&lt;p&gt;-------------------------------------------------&lt;br /&gt;
CONVERT *.MP3 to *.GSM&lt;/p&gt;
&lt;p&gt;&lt;/code&gt;&lt;br /&gt;
for I in $(ls -1 *.mp3)&lt;br /&gt;
do&lt;br /&gt;
        NAME=$I&lt;br /&gt;
        echo "Converting from mp3.wav"&lt;br /&gt;
        mpg123 -w "$NAME.wav" "$NAME"&lt;br /&gt;
        echo "Converting from wav to gsm"&lt;br /&gt;
        sox -t wav "$NAME.wav" -r 8000 -c1 -t gsm "$NAME.gsm" resample -ql&lt;br /&gt;
        echo "output filename: $NAME.gsm"&lt;br /&gt;
done&lt;br /&gt;
&lt;/code&gt;&lt;/p&gt;
&lt;p&gt;Not sure if this works when there are spaces in the file names...&lt;/p&gt;
</description>
 <pubDate>Mon, 14 Jan 2008 20:22:15 -0800</pubDate>
</item>
<item>
 <title>Asterisk Click 2 Call Application</title>
 <link>http://www.ctunion.com/node/176</link>
 <description>&lt;p&gt;I found a great link for how to do a click to call implementation on the web site.  &lt;/p&gt;
&lt;p&gt;&lt;a href="http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html"&gt; http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html &lt;/a&gt;&lt;/p&gt;
&lt;p&gt;I was able to get it working in under an hour!&lt;/p&gt;
</description>
 <pubDate>Mon, 26 Feb 2007 12:42:30 -0800</pubDate>
</item>
<item>
 <title>How to fix "Unable to handle indication 3 for" / Asterisk Error Message</title>
 <link>http://www.ctunion.com/node/137</link>
 <description>&lt;p&gt;How to fix "Unable to handle indication 3 for" / Asterisk Error Message&lt;/p&gt;
&lt;p&gt;The other day i had set up a new asterisk box and wanted to start making some calls.  I noticed that anytime a SIP = SIP call was made, no ringing would be heard on the callers end.  Additionally it would be accompanied by an error message about "indication 3"&lt;/p&gt;
&lt;p&gt;Here is how you fix it...&lt;/p&gt;
&lt;p&gt;1) cd /etc/asterisk&lt;br /&gt;
2) nano indications.conf&lt;br /&gt;
3) paste the following code into the conf file...&lt;/p&gt;
&lt;p&gt;&lt;code &gt;&lt;br /&gt;
;&lt;br /&gt;
 ;  Static indications configuration files, used by&lt;br /&gt;
 ; the pbx_indications module.&lt;br /&gt;
 ;&lt;br /&gt;
 ; The "general" category is for certain variables.  All other categories&lt;/p&gt;
</description>
 <pubDate>Sun, 20 Aug 2006 12:08:42 -0700</pubDate>
</item>
<item>
 <title>Get Festival Working in Asterisk on Debian after you compile Asterisk from source</title>
 <link>http://www.ctunion.com/node/134</link>
 <description>&lt;p&gt;1) apt-get install festival&lt;br /&gt;
2) nano /etc/festival.scm&lt;br /&gt;
3) paste:&lt;/p&gt;
&lt;p&gt;&lt;code &gt;&lt;br /&gt;
;;; Command for Asterisk begin&lt;br /&gt;
(define (tts_textasterisk string mode)&lt;br /&gt;
    "(tts_textasterisk STRING MODE)&lt;br /&gt;
  Apply tts to STRING.  This function is specifically designed for&lt;br /&gt;
  use in server mode so a single function call may synthesize the string.&lt;br /&gt;
  This function name may be added to the server safe functions."&lt;br /&gt;
      (utt.send.wave.client (utt.wave.resample (utt.wave.rescale (utt.synth&lt;br /&gt;
                    (eval (list 'Utterance 'Text string))) 5) 8000)))&lt;br /&gt;
;;; Command for Asterisk end&lt;br /&gt;
&lt;/code&gt;&lt;br /&gt;
4) restart festival / restart asterisk&lt;/p&gt;
</description>
 <pubDate>Sat, 05 Aug 2006 20:05:50 -0700</pubDate>
</item>
<item>
 <title>A Marriage Made in Heaven: Sprint/Nextel Cellphone + Asterisk = Unlimited U.S. Cell Phone Calls for $5</title>
 <link>http://www.ctunion.com/node/127</link>
 <description>&lt;p&gt;Huray! VOIP.&lt;/p&gt;
&lt;p&gt;Sprint announced the availability of a new add-on for existing and new Sprint cellphone customers. For $5 more a month and a little Yankee ingenuity, you now can make unlimited FREE calls between your Sprint cellphone (or multiple PCS phones if you’re on a shared plan) and your residential phone number regardless of the wireline carrier. In short, your home phone service need not be with Sprint. If you have Sprint home phone service, then the new PCS to Home service will be free. In either case, no cellphone minutes will be assessed for inbound (read comments!) or outbound calls between your Sprint cellphone and your home number … ever. In fact, they’ll show up on your statement as PCS-to-PCS calls which are also free.&lt;/p&gt;
</description>
 <pubDate>Sat, 22 Jul 2006 16:15:41 -0700</pubDate>
</item>
<item>
 <title>Setting up Asterisk and E911</title>
 <link>http://www.ctunion.com/node/118</link>
 <description>&lt;p&gt;The Asterisk Open Source PBX provides a cost effective way of achieving E911 compatibility when used in conjunction with a telephone carrier that offers E911 database management services to its customers.   This database management service consists of a reverse directory maintained by the customer, but residing with the carrier, that assigns location information to private (DID) telephone numbers that the customer uses.  &lt;/p&gt;
&lt;p&gt;In order for E911 to function on an Asterisk PBX, each extension phone must be assigned a 10-digit DID number.  This DID number is then registered with the telephone carrier along with the location of the phone.&lt;/p&gt;
</description>
 <pubDate>Mon, 29 May 2006 18:33:02 -0700</pubDate>
</item>
<item>
 <title>Asterisk Day / Night Mode Example</title>
 <link>http://www.ctunion.com/node/113</link>
 <description>&lt;p&gt;exten =&amp;gt; 4710,1,GotoIfTime(17:00-8:59|*|*|*?night-mode|1) ;break at night send directly to message otherwise just continue and dial&lt;br /&gt;
  exten =&amp;gt; 4710,2,GotoIfTime(12:00-12:59|*|*|*?night-mode|1);break for lunch&lt;br /&gt;
  exten =&amp;gt; 4710,3,Dial(Sip/521,15,tr)&lt;br /&gt;
  exten =&amp;gt; 4710,4,Goto(night-mode,1)&lt;/p&gt;
&lt;p&gt;  exten =&amp;gt; night-mode,1,Answer&lt;br /&gt;
  exten =&amp;gt; night-mode,2,Wait(2)&lt;br /&gt;
  exten =&amp;gt; night-mode,3,SetMusicOnHold(default)&lt;br /&gt;
  exten =&amp;gt; night-mode,4,DigitTimeout,3&lt;br /&gt;
  exten =&amp;gt; night-mode,5,ResponseTimeout,20&lt;br /&gt;
  exten =&amp;gt; night-mode,6,Background(/astsounds/p1)&lt;br /&gt;
  exten =&amp;gt; night-mode,7,SetVar(LISTCOUNT=$[${LISTCOUNT} + 1]) ; increment&lt;/p&gt;
</description>
 <pubDate>Thu, 11 May 2006 01:04:48 -0700</pubDate>
</item>
<item>
 <title>A sample Dialplan for an accounting office</title>
 <link>http://www.ctunion.com/node/48</link>
 <description>&lt;p&gt;Hey I just wanted to share with everyone the first dialplan that i created for a small 2 person accounting office.  Have fund&lt;/p&gt;
&lt;p&gt;&lt;code &gt;&lt;br /&gt;
[aa_1]&lt;br /&gt;
exten =&amp;gt; 5,1,Playtones(!980/230,0)&lt;br /&gt;
exten =&amp;gt; 5,2,AGI(festival-script.pl|Take 8 East then take the 67 north to wintergardens south exit.)&lt;br /&gt;
exten =&amp;gt; 5,3,AGI(festival-script.pl|Travel approximatly 1 point 5 miles to. eight. eight. zero. seven. wintergardens boulevard.)&lt;br /&gt;
exten =&amp;gt; 5,4,AGI(festival-script.pl|You will see us on the left hand side.  The closest cross street is Garde-nia.)&lt;br /&gt;
exten =&amp;gt; 5,5,Wait(2)&lt;br /&gt;
exten =&amp;gt; 5,6,Background(to-hear-msg-again)&lt;br /&gt;
exten =&amp;gt; 5,7,Background(press)&lt;/p&gt;
</description>
 <pubDate>Wed, 02 Nov 2005 16:01:49 -0800</pubDate>
</item>
<item>
 <title>Books Available Amazon.com related to asterisk programming</title>
 <link>http://www.ctunion.com/node/37</link>
 <description>&lt;p&gt;So you want to learn more about asterisk PBX programming and are thinking about getting a book to help you. I have researched this and as of this posting, there are currently 2 books available on the subject.  I went ahead and bought them both.  &lt;/p&gt;
&lt;p&gt;The first is a bright yellow book titled "VOIP Telephony with Asterisk.  A technical overview of the open source PBX" written by Paul Mahler.  &lt;/p&gt;
&lt;p&gt;The second is a white book with a starfish on the cover titled "Asterisk the future of telephony" written by Jim Van Meggelen, Jared Smith &amp;amp; Leif Madsen&lt;/p&gt;
&lt;p&gt;First let me tell you about the yellow book by Paul Mahler.  In my opinion, this guy is basically throwing together a book with little understanding of asterisk or how to actually use it.  He is basically copy and pasting together several different manuals into chapters and calling it a book.  The book does a good job of describing the very basic functions of asterisk and goes over a few of the concepts but that’s about it.  Don't get this book if you want to do anything more complex then get 3 phones calling each other.  Also I would say that before I got this book I had spent considerable time online doing research and was getting frustrated with certain topics being too complex.  This book had not gotten past what I learned the first day reading online.&lt;/p&gt;
</description>
 <pubDate>Thu, 20 Oct 2005 21:16:12 -0700</pubDate>
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