<?xml version="1.0" encoding="utf-8" standalone="yes"?><rss version="2.0" xmlns:atom="http://www.w3.org/2005/Atom" xmlns:content="http://purl.org/rss/1.0/modules/content/"><channel><title>Asterisk-Pbx on Jackenhack</title><link>https://www.jackenhack.com/categories/asterisk-pbx/</link><description>Recent content in Asterisk-Pbx on Jackenhack</description><generator>Hugo</generator><language>en-us</language><lastBuildDate>Tue, 06 Jun 2006 00:00:00 +0000</lastBuildDate><atom:link href="https://www.jackenhack.com/categories/asterisk-pbx/index.xml" rel="self" type="application/rss+xml"/><item><title>JackenIAX with multiple lines support</title><link>https://www.jackenhack.com/jackeniax-with-multiple-lines-support/</link><pubDate>Tue, 06 Jun 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/jackeniax-with-multiple-lines-support/</guid><description>&lt;p&gt;I finally completed a new version of JackenIAX. It now has up to four lines support. I also added a searchable call log. Lot&amp;rsquo;s of other small improvements as well.&lt;/p&gt;</description></item><item><title>JackenIAX and multiple lines</title><link>https://www.jackenhack.com/jackeniax-and-multiple-lines/</link><pubDate>Sun, 19 Mar 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/jackeniax-and-multiple-lines/</guid><description>&lt;p&gt;I finally have a version of JackenIAX that supports multiple lines. I have added so you can choose to output background music for the calls on hold. There&amp;rsquo;s a lot of testing, and I also have to rethink the user interface to be able to handle multiple calls. But i think it&amp;rsquo;s going to be cool.&lt;/p&gt;</description></item><item><title>Adding new features</title><link>https://www.jackenhack.com/adding-new-features/</link><pubDate>Wed, 08 Mar 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/adding-new-features/</guid><description>&lt;p&gt;Every time I answer the phone (yes, I&amp;rsquo;m using JackenIAX), I have to pause iTunes before answering. So today I added an option so when you answer or dial, iTunes pauses automatically. It will be in the next release (within a couple of days) I had to rewrite a large section of code for some other features that I have planned. I&amp;rsquo;m getting a bit more comfortable with Cocoa programming, and discovers old code full of brain farts. And the code is getting smaller with more features. A step in the right direction&amp;hellip;&lt;/p&gt;</description></item><item><title>New version of JackenIAX</title><link>https://www.jackenhack.com/new-version-of-jackeniax/</link><pubDate>Mon, 06 Mar 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/new-version-of-jackeniax/</guid><description>&lt;p&gt;I just put up a new version of JackenIAX. Added Applescript support, URL dialing (currently only via Applescript), DTMF tones when dialing etc.&lt;/p&gt;</description></item><item><title>Looks like it works on Intel Mac</title><link>https://www.jackenhack.com/looks-like-it-works-on-intel-mac/</link><pubDate>Sun, 26 Feb 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/looks-like-it-works-on-intel-mac/</guid><description>&lt;p&gt;&lt;img alt="Jackeniax Icon" decoding="async" height="126" loading="lazy" src="https://www.jackenhack.com/looks-like-it-works-on-intel-mac/images/jackeniax_icon1_hu_9f3cd4413abe4492.png" width="131"&gt; I just received a mail from a user of JackenIAX. He&amp;rsquo;s running a patched version of Mac OS X on a generic PC, and now the program works on his machine. Cool! So now I guess JackenIAX is working on Intel Macs as well. Debugging via proxy (I don&amp;rsquo;t have an Intel Mac) is not the way to go. You could press the Donate button on this page, and I promise to buy a Intel Mac for debugging. Nudge Nudge&amp;hellip;&lt;/p&gt;</description></item><item><title>New features in JackenIAX</title><link>https://www.jackenhack.com/new-features-in-jackeniax/</link><pubDate>Thu, 23 Feb 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/new-features-in-jackeniax/</guid><description>&lt;p&gt;I&amp;rsquo;ve added a couple of new features in JackenIAX.&lt;/p&gt;
&lt;p&gt;Added a symbol to indicate when you got new voice messages in your Asterisk server. Rewrote all of the code for the indicator symbols. Nothing you will notice, but I feel good about it. Now the codec selection actually work. Input validation of all preferences.&lt;/p&gt;</description></item><item><title>Problems on the other end</title><link>https://www.jackenhack.com/problems-on-the-other-end/</link><pubDate>Mon, 20 Feb 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/problems-on-the-other-end/</guid><description>&lt;p&gt;As you may know I recently released my new program for VoIP for the mac. I have written instructions on the home-page how to get a free account at FWD Free World Dialup to test. It seems they have a lot of downtime, so I get a lot of bug reports. I really need another free IAX provider to recommend. Anyone knows about one?&lt;/p&gt;</description></item><item><title>First beta of JackenIAX released</title><link>https://www.jackenhack.com/first-beta-of-jackeniax-released/</link><pubDate>Sat, 18 Feb 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/first-beta-of-jackeniax-released/</guid><description>&lt;p&gt;&lt;img alt="Jackeniax Icon" decoding="async" height="126" loading="lazy" src="https://www.jackenhack.com/first-beta-of-jackeniax-released/images/jackeniax_icon_hu_9f3cd4413abe4492.png" width="131"&gt; So it&amp;rsquo;s finally time to release my IAX softphone for Macintosh. It&amp;rsquo;s still very much in beta, but I need feedback on how everything works. There&amp;rsquo;s a lot of stuff that I&amp;rsquo;m adding that didn&amp;rsquo;t make it into this release. Have a look here.&lt;/p&gt;</description></item><item><title>The release of the first beta of JackenIAX SoftPhone for Mac OS 10.4 is imminent</title><link>https://www.jackenhack.com/the-release-of-the-first-beta-of-jackeniax-softphone-for-mac-os-104-is-imminent/</link><pubDate>Fri, 17 Feb 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/the-release-of-the-first-beta-of-jackeniax-softphone-for-mac-os-104-is-imminent/</guid><description>&lt;p&gt;I just need to write rudimentary install instructions, finish the simple homepage for downloading and I&amp;rsquo;m in business. So any day now&amp;hellip;&lt;/p&gt;
&lt;p&gt;If you are in a hurry, mail me at &lt;a href="mailto:info@jackenhack.com"&gt;info@jackenhack.com&lt;/a&gt; and ask if you can join the beta team.&lt;/p&gt;</description></item><item><title>Waiting time</title><link>https://www.jackenhack.com/waiting-time/</link><pubDate>Thu, 19 Jan 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/waiting-time/</guid><description>&lt;p&gt;I haven&amp;rsquo;t had time to update my blog for a while. The last couple of days has been spent trying to hunt down an annoying bug in my IAX client I&amp;rsquo;m writing. Sometimes when dialing in, my program would not get a notification from the Asterisk server that a call was in progress. So I have tested all sorts of things to try to find the bug. So today I finally used tcpdump to see exactly what kind of packets I got from the server. And the problem was that I connected from the inside of my firewall, out trough and back again! So it was just an miss-configuration, the program has been working all along. Isn&amp;rsquo;t the Darwin award coming up soon?&lt;/p&gt;</description></item><item><title>VU Meter</title><link>https://www.jackenhack.com/vu-meter/</link><pubDate>Mon, 02 Jan 2006 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/vu-meter/</guid><description>&lt;p&gt;&lt;img alt="Vumeter" decoding="async" height="92" loading="lazy" src="https://www.jackenhack.com/vu-meter/images/vumeter_hu_e967c9a3fcc2c934.png" width="160"&gt;&lt;/p&gt;
&lt;p&gt;I&amp;rsquo;ve been hard at work on my IAX client software the last couple of days. I also have re-written a lot of the code. I decided early on that I wanted to have classical VU-meters to display the in and output volume. But I don&amp;rsquo;t want them to take to much CPU time. So I made a lot of optimizations on that code. So after the rewrite I managed to lower the CPU utilisation by half. And it&amp;rsquo;s fun to do programming in Cocoa, but the framework is hard to learn by the volume of it. But slowly I&amp;rsquo;m getting more comfortable.&lt;/p&gt;</description></item><item><title>Compiling</title><link>https://www.jackenhack.com/compiling/</link><pubDate>Wed, 23 Nov 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/compiling/</guid><description>&lt;p&gt;The IAX client I&amp;rsquo;m writing is coming along nicely. It&amp;rsquo;s now in a state that makes the program usable. And today I made changes so I could compile a universal binary. And the program works on my PPC mac, but I use some libraries I needed to change the makefiles for, so I don&amp;rsquo;t know if they work on a Intel Mac. It would be cool if I could get the first release to support universal binaries, so hopefully I can find someone who has a developer machine, or a pc with a hacked Mac OS X for Intel on it.&lt;/p&gt;</description></item><item><title>Setting up SUBSCRIBE/NOTIFY BLF in Asterisk@Home for Grandstream GXP-2000 phones</title><link>https://www.jackenhack.com/setting-up-subscribenotify-blf-in-asteriskhome-for-grandstream-gxp-2000-phones/</link><pubDate>Tue, 22 Nov 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/setting-up-subscribenotify-blf-in-asteriskhome-for-grandstream-gxp-2000-phones/</guid><description>&lt;p&gt;With the latest beta firmware in the &lt;a href="http://www.amazon.com/gp/product/B001ROZZ44?ie=UTF8&amp;amp;tag=jackensweblog-20&amp;amp;linkCode=as2&amp;amp;camp=1789&amp;amp;creative=390957&amp;amp;creativeASIN=B001ROZZ44"&gt;Grandstream GXP-2000 VoIP phone&lt;/a&gt;, the have added support for BLF. This means that you can use the quick dial LEDs for status of extensions. You can monitor all extensions on your phone. If a extension is busy, the led for that extension is lit. You can also dial directly to the extension by pressing the dial button. Very handy.&lt;/p&gt;</description></item><item><title>Application name</title><link>https://www.jackenhack.com/application-name/</link><pubDate>Thu, 10 Nov 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/application-name/</guid><description>&lt;p&gt;I now have a working IAX soft phone up and running. Answering and dialing works. The sound quality is good. And the amount of cpu usage is around three percent. But as soon as I add all the animating graphics, like sound input/output meters and stuff, that figure will go up substantially. But one of the goals is to try to avoid letting the program becoming a resource hog.&lt;/p&gt;
&lt;p&gt;I decided to really try to stay focused on the basic stuff to really iron out the bugs/memory leaks before continuing to the fun stuff like user interface. But thanks to Project Builder, it&amp;rsquo;s easy to make a usable interface immediately.&lt;/p&gt;</description></item><item><title>IAX Soft Phones for Mac OS X</title><link>https://www.jackenhack.com/iax-soft-phones-for-mac-os-x/</link><pubDate>Tue, 08 Nov 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/iax-soft-phones-for-mac-os-x/</guid><description>&lt;p&gt;I&amp;rsquo;ve been looking for a softphone for Macintosh that uses IAX, but I haven&amp;rsquo;t been able to find a good one. The &lt;a href="http://www.xten.com/index.php?menu=eyeBeam" title="EyeBeam SIP Soft phone"&gt;EyeBeam&lt;/a&gt; and &lt;a href="http://www.xten.com/index.php?menu=X-Series" title="X-lite SIP Soft phone"&gt;X-lite&lt;/a&gt; uses sip and has stability problems so they are not an option either. So I started to write my own softphone. It&amp;rsquo;s nice to be programming in &lt;a href="http://developer.apple.com/cocoa/" title="Cocoa Object oriented Application environment"&gt;Cocoa&lt;/a&gt; again. But I&amp;rsquo;m becoming old, so I have to read up on programming in Cocoa again. I have a working softphone up and running already. Well, sort of. It receives calls and you can answer and talk. But I&amp;rsquo;ve only been writing on it for two days, and most of the time has been spent on installing a Subversion system on my server. As soon as I get a polished version, I&amp;rsquo;ll put up a beta program. I&amp;rsquo;m thinking of calling the program &amp;ldquo;Breaker-Breaker!&amp;rdquo; in the old CB-radio spirit. And yes, it will have the old style analog VU indicators for bandwidth and voice volume.&lt;/p&gt;</description></item><item><title>Gizmo Project</title><link>https://www.jackenhack.com/gizmo-project/</link><pubDate>Sun, 23 Oct 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/gizmo-project/</guid><description>&lt;p&gt;&lt;img alt="Gizmo-Logo2" decoding="async" height="90" loading="lazy" src="https://www.jackenhack.com/gizmo-project/images/Gizmo-logo2.gif" width="237"&gt; As soon as you mention the word VoIP, everyone responds with &amp;ldquo;Skype?&amp;rdquo; This immediately gets me into an discussion why people should not use Skype. The fact that your computer is used as a link to route traffic, much like BitTorrent. That it is based on a proprietary protocol, only known by Skype, so no other manufacturers can make compatible equipment without permisson (and probably a big fat wad of cash) to Skype Corporation. All this means that there&amp;rsquo;s no connection between Skype and regular SIP phones that is emerging as the standard for VoIP calls. But all hope is not lost.&lt;/p&gt;</description></item><item><title>Problem dialing in to Asterisk@Home from FWD</title><link>https://www.jackenhack.com/problem-dialing-in-to-asteriskhome-from-fwd/</link><pubDate>Sun, 16 Oct 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/problem-dialing-in-to-asteriskhome-from-fwd/</guid><description>&lt;p&gt;I helped my friend Micke to set up an Asterisk@Home PBX server today. But when setting up Free World Dialup FWD, we had some problems. First I forgot to &lt;a href="http://www.siptosip.net"&gt;enable IAX in the account setup&lt;/a&gt;. After that, dialing out worked, but we could not dial in to his FWD number. To connect we use an IAX trunk. We searched for an hour trying to find the problem, and after searching Google we finally found the answer. In the trunk settings for FWD, the incoming user context has to be named &lt;strong&gt;iaxfwd&lt;/strong&gt;. Irritating to say the least. After changing the name everything worked fine.&lt;/p&gt;</description></item><item><title>Free book about Asterisk</title><link>https://www.jackenhack.com/free-book-about-asterisk/</link><pubDate>Sat, 15 Oct 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/free-book-about-asterisk/</guid><description>&lt;p&gt;&lt;img alt="Asterisk book" decoding="async" height="128" loading="lazy" src="https://www.jackenhack.com/free-book-about-asterisk/images/asterisk_book_cover_tn_hu_d63a55bcc2434f35.jpg" width="128"&gt;Asterisk: The Future of Telephony online from O&amp;rsquo;Reilly Media. They have released it under the &lt;a href="http://creativecommons.org/"&gt;Creative Commons license&lt;/a&gt;. &lt;a href="http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11"&gt;This means that you can download the book here for free&lt;/a&gt;. But it also means that you should &lt;a href="http://www.amazon.com/exec/obidos/redirect?link_code=ur2&amp;amp;camp=1789&amp;amp;tag=jackensweblog-20&amp;amp;creative=9325&amp;amp;path=http://www.amazon.com/gp/product/0596009623?v=glance%26n=283155%26n=507846%26s=books%26v=glance"&gt;buy the book here.&lt;/a&gt; &lt;img decoding="async" loading="lazy" src="https://www.jackenhack.com/free-book-about-asterisk/images/ir"&gt;Now you have the downloaded pdf version for quick searching, and the nice feel of a paper copy for browsing.&lt;/p&gt;</description></item><item><title>A professional sounding Digital Receptionist in Asterisk</title><link>https://www.jackenhack.com/setting-up-a-professional-sounding-digital-receptionist-in-asteriskhome/</link><pubDate>Fri, 14 Oct 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/setting-up-a-professional-sounding-digital-receptionist-in-asteriskhome/</guid><description>&lt;p&gt;&lt;img alt="Asterisk" decoding="async" height="96" loading="lazy" src="https://www.jackenhack.com/setting-up-a-professional-sounding-digital-receptionist-in-asteriskhome/images/asterisk.gif" width="174"&gt; After setting up my Asterisk@Home server and do all the configuring to get all lines and extensions up and running, it was time for recording the Digital Receptionist messages. My first stab at it wasn&amp;rsquo;t that great, let&amp;rsquo;s face it, I&amp;rsquo;m no voice-over announcer. I wanted a little more professionally sounding recording. So I started to search the net for a way to do it, and I think I found a pretty good solution.&lt;/p&gt;</description></item><item><title>SIP Voip Phone Grandstream GXP-2000 Review</title><link>https://www.jackenhack.com/sip-voip-phone-grandstream-gxp-2000-review/</link><pubDate>Sat, 08 Oct 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/sip-voip-phone-grandstream-gxp-2000-review/</guid><description>&lt;p&gt;&lt;img alt="Grandstream voip phone" decoding="async" height="150" loading="lazy" src="https://www.jackenhack.com/sip-voip-phone-grandstream-gxp-2000-review/images/GXP2000_front_p_hu_676c9d6861d18f9.jpg" title="Grandstream voip phone" width="240"&gt;&lt;/p&gt;
&lt;p&gt;I got my first real Voip phone, the &lt;a href="http://www.grandstream.com"&gt;Grandstream GXP-2000&lt;/a&gt; a week ago.&lt;/p&gt;
&lt;p&gt;I have two ATA boxes to connect my old wireless phones, but I wanted to try out a real IP-phone. It&amp;rsquo;s a nice enough phone, feels a bit cheap (which it is), but I guess you can&amp;rsquo;t get everything for $100. Connecting it to my &lt;a href="http://www.asterisk.org/" title="Asterisk Open Source PBX Software"&gt;Asterisk&lt;/a&gt; server was a no-brainer. It has a web interface for setup, but you can do basic setup like ip-numbers from the phonepad. It&amp;rsquo;s great to have an message-waiting led on the phone. Then you just press the MSG button to listen to your voice-box messages. To enable this function, just put *97 in the field &amp;ldquo;Voice Mail UserID&amp;rdquo; in the Account page of the web setup. There&amp;rsquo;s support for up to 11 line indicators (with an additional key-module) but I think I&amp;rsquo;ll do fine with four. The only thing I haven&amp;rsquo;t got working is the Conference button, but it&amp;rsquo;s easy to just transfer the calls to a conference number. You also have 7 quick dial buttons. To configure them, just go to the web setup for the phone and enter the name and number. The sound quality is excellent. Much better than both the &lt;a href="http://www.amazon.com/exec/obidos/redirect?link_code=ur2&amp;amp;camp=1789&amp;amp;tag=jackensweblog-20&amp;amp;creative=9325&amp;amp;path=tg/detail/-/B0002V8KX6/qid=1128813530/sr=8-1/ref=pd_bbs_1?v=glance%26s=pc%26n=507846"&gt;Linksys PAP2&lt;/a&gt;&lt;img decoding="async" loading="lazy" src="https://www.jackenhack.com/sip-voip-phone-grandstream-gxp-2000-review/images/ir"&gt; and the &lt;a href="http://www.i3micro.com/"&gt;i3 micro Vood VRG-121&lt;/a&gt; box I have for my old analog phones. The speaker phone works, but isn&amp;rsquo;t fantastic. I normally just use the speaker phone while waiting for the call to go through, so for me that&amp;rsquo;s not a problem. There&amp;rsquo;s a new firmware I&amp;rsquo;ve just installed that&amp;rsquo;s supposed to make the speaker phone part work better, but I haven&amp;rsquo;t had time to test it fully. So would I recommend this phone? Yes, if you need a cheap voip phone that uses SIP I can really recommend this one.&lt;/p&gt;</description></item><item><title>Adding blacklist to an Asterisk PBX Server</title><link>https://www.jackenhack.com/adding-blacklist-to-an-asteriskhome-pbx-voip-server/</link><pubDate>Mon, 26 Sep 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/adding-blacklist-to-an-asteriskhome-pbx-voip-server/</guid><description>&lt;p&gt;&lt;strong&gt;Update:&lt;/strong&gt; The code below does not work in Asterisk 2.0b4 or newer. I will update with instructions on how to get it to work in those versions. Thanks to &lt;a href="http://www.thestroots.com"&gt;Dan&lt;/a&gt; for pointing this out.&lt;/p&gt;
&lt;p&gt;&lt;strong&gt;Update:&lt;/strong&gt; I have changed the code to the removed part. It now requires a key press to accept the number to delete.&lt;/p&gt;
&lt;p&gt;I have added a blacklist function to my Asterisk@Home server and thought that I&amp;rsquo;d share it here if someone is interested. There are two ways to add a number to the blacklist. The easiest is to dial *32. Then you get a voice telling you the number of the last incoming call. To add it to the blacklist, press 1. Great use against pesky telemarketers and If you want to add it manually,  press *30. Then you can enter the phone number directly on your telephone. To remove an entry from the blacklist, press *31 and enter the phone number. If you don&amp;rsquo;t like the codes I use, change them.&lt;/p&gt;</description></item><item><title>Adding ENUM support in Asterisk@Home</title><link>https://www.jackenhack.com/adding-enum-support-in-asteriskhome/</link><pubDate>Tue, 13 Sep 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/adding-enum-support-in-asteriskhome/</guid><description>&lt;p&gt;There&amp;rsquo;s a system called ENUM that enables VoIP servers like Asterisk to check the number your dialing to see if there is a possibility to connect directly to your server instead of going over PSTN land lines. So I registered at E164.org and added my phone number. I tried to figure out how to add this functionality to Asterisk@Home, but couldn&amp;rsquo;t find anything on how to do it. But I finally managed to get it to work, and it&amp;rsquo;s really simple. Here&amp;rsquo;s how I did it.&lt;/p&gt;</description></item><item><title>Screaming monkeys</title><link>https://www.jackenhack.com/screaming-monkeys/</link><pubDate>Tue, 13 Sep 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/screaming-monkeys/</guid><description>&lt;p&gt;Today I finally got to see a tele-marketer getting caught in my Asterisk server. I&amp;rsquo;ve made a blacklist system, so I can blacklist callers. As soon as they call, they get transfered to a voice saying &amp;ldquo;There is no one to take your call, they&amp;rsquo;ve all been carried away by monkeys&amp;rdquo; and then a lot of monkeys screaming. And the dumb fuck called back three times before giving up. One down, three million to go&amp;hellip;&lt;/p&gt;</description></item><item><title>More Asterisk AGI</title><link>https://www.jackenhack.com/more-asterisk-agi/</link><pubDate>Fri, 09 Sep 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/more-asterisk-agi/</guid><description>&lt;p&gt;I have finally finished my first AGI program for my Asterisk server. So time for my next program. I have installed &lt;a href="http://www.nagios.org/"&gt;Nagios network monitoring program&lt;/a&gt; on one of my machines. I want Nagios to call my mobile phone when one of my servers go down and using Festival, giving me a voice message of what server and the type of problem.&lt;/p&gt;</description></item><item><title>Linux is taking over my home!</title><link>https://www.jackenhack.com/linux-is-taking-over-my-home/</link><pubDate>Sun, 04 Sep 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/linux-is-taking-over-my-home/</guid><description>&lt;p&gt;I&amp;rsquo;ve had performance problems with my firewall so I thought it was time to upgrade to a better firewall. After looking around I found a lot of articles about the Linksys WRT54G, a wireless router that&amp;rsquo;s easily patched with new firmware. Because it uses the linux operating system, Linksys have to publish source code under the GPL license. This makes it easy for others to improve and implement new features. So I got an image from &lt;a href="http://www.sveasoft.com/"&gt;Sveasoft&lt;/a&gt; called &lt;a href="http://www.sveasoft.com/content/view/20/1/"&gt;Talisman&lt;/a&gt; and installed it. And it works great! I can now have &lt;a href="http://en.wikipedia.org/wiki/QOS"&gt;QoS&lt;/a&gt; so my VoIP telephone stuff always gets priority.&lt;/p&gt;</description></item><item><title>Continued adventures with Asterisk</title><link>https://www.jackenhack.com/continued-adventures-with-asterisk/</link><pubDate>Sat, 20 Aug 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/continued-adventures-with-asterisk/</guid><description>&lt;p&gt;As usual when I find something new, it goes a bit overboard. I always wanted to delve into learning the Perl programming language. So I took this opportunity to use it to write a small plug-in for my Asterisk server. As soon as somebody calls, my script goes to a Swedish directory service and looks up the name. It then puts it in the callerID information so the name shows up on my phones. It also connects to my satellite receiver and displays a message on my tv so I know who is calling. Pretty nifty. Naturally I found that someone already written this in php, and that they&amp;rsquo;ve used the wap version of the lookup page to find the name. That preserves a lot of bandwidth so I&amp;rsquo;ll probably rewrite my program to use that instead. Anyway, Perl is an amazing language.&lt;/p&gt;</description></item><item><title>Asterisk PBX</title><link>https://www.jackenhack.com/asterisk-pbx/</link><pubDate>Sun, 14 Aug 2005 00:00:00 +0000</pubDate><guid>https://www.jackenhack.com/asterisk-pbx/</guid><description>&lt;p&gt;&lt;img alt="Asterisk" decoding="async" height="96" loading="lazy" src="https://www.jackenhack.com/asterisk-pbx/images/asterisk.gif" width="174"&gt;&lt;/p&gt;
&lt;p&gt;So I&amp;rsquo;ve been busy for the last five days setting up my &lt;a href="http://www.asterisk.org/"&gt;Asterisk PBX&lt;/a&gt;. I&amp;rsquo;ve connected two different VoIP lines to it. I have the &lt;a href="http://www.xten.com/"&gt;X-Lite&lt;/a&gt; VoIP softphones installed on all the computers and an FSX device so I can use my regular wireless phone. And it&amp;rsquo;s enjoyable learning something new. First, my wife thought that I was crazy, &amp;ldquo;Who needs a telephone switchboard at home!?&amp;rdquo; But I think she&amp;rsquo;s coming around. Now she can sit at her computer using a headset and talk to her friends without holding the line. And it&amp;rsquo;s easy to transfer calls between all the phones in the apartment. It&amp;rsquo;s also nice to have a digital answering machine and get the messages mailed to you. The added benefit of multiple lines on the same phone number and that it doesn&amp;rsquo;t cost anything to dial a phone in Sweden is a bonus.&lt;/p&gt;</description></item></channel></rss>