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	<title>
	Comments on: Newbie&#8217;s Guide to Asterisk@Home 2.5: Unabridged Soup-to-Nuts Installation Guide	</title>
	<atom:link href="https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/feed/" rel="self" type="application/rss+xml" />
	<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/</link>
	<description>Ward Mundy&#039;s Technobabblelog</description>
	<lastBuildDate>Thu, 02 Jun 2011 14:13:55 +0000</lastBuildDate>
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	<item>
		<title>
		By: Matthew		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1733</link>

		<dc:creator><![CDATA[Matthew]]></dc:creator>
		<pubDate>Tue, 11 Jul 2006 16:06:07 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1733</guid>

					<description><![CDATA[This site is one-of-a-kind! I have never come across any site better than this. What are the odds of adding IM (Instant Messaging) to it? 

Again..Well Done!]]></description>
			<content:encoded><![CDATA[<p>This site is one-of-a-kind! I have never come across any site better than this. What are the odds of adding IM (Instant Messaging) to it? </p>
<p>Again..Well Done!</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Craig		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1662</link>

		<dc:creator><![CDATA[Craig]]></dc:creator>
		<pubDate>Tue, 20 Jun 2006 01:19:02 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1662</guid>

					<description><![CDATA[I can&#039;t get X-Lite to register with my Asterisk Server.  The internal IP address for Asterisk.  Everything works fine when I put the internal IP address with X-Lite; however, when I attempt to use my external static IP address, X-Lite will not work.  What am I doing wrong?

Thanks,
Craig

&lt;i&gt;[WM: You can&#039;t get to your own external IP address from inside with most NAT-based routers.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>I can&#8217;t get X-Lite to register with my Asterisk Server.  The internal IP address for Asterisk.  Everything works fine when I put the internal IP address with X-Lite; however, when I attempt to use my external static IP address, X-Lite will not work.  What am I doing wrong?</p>
<p>Thanks,<br />
Craig</p>
<p><i>[WM: You can&#8217;t get to your own external IP address from inside with most NAT-based routers.]</i></p>
]]></content:encoded>
		
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		<item>
		<title>
		By: therock		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1143</link>

		<dc:creator><![CDATA[therock]]></dc:creator>
		<pubDate>Tue, 07 Mar 2006 15:58:35 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1143</guid>

					<description><![CDATA[Fantastic article. Ever since I stepped into the voip telephony world, Ward and family have become my new best friends. Thank you for taking the time to write these articles, you are simply fantastic.

I have setup a nice Asterisk @home system, I got a stealth autoattendant going. I have implemented the ivr which says press 1 to continue which then gets forwarded to a ring group which rings all extensions and then gets dumped to an extensions vmail. If 1 is not received from the caller, then it is assumed this could be a telemarketer and a Zapateller sit tone is sent out, I am plagued with telemarketers!! My problem is, when the caller presses 1, i want them to hear music while all the phones ring, but i am not able to get the music on. Any ideas?

my extensions_additional.conf ivr code:

[aa_1]
include =&gt; aa_1-custom
exten =&gt; 1,1,Goto(ext-group,1,1)	; jump
exten =&gt; fax,1,Goto(ext-fax,in_fax,1)
exten =&gt; h,1,Hangup
exten =&gt; hang,1,Playback(vm-goodbye)
exten =&gt; hang,2,Hangup
exten =&gt; i,1,Playback(invalid)
exten =&gt; i,2,Goto(s,7)
include =&gt; ext-local
include =&gt; app-messagecenter
include =&gt; app-directory
exten =&gt; s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4)
exten =&gt; s,2,Answer
exten =&gt; s,3,Wait(1)
exten =&gt; s,4,SetVar(LOOPED=1)
exten =&gt; s,5,GotoIf($[${LOOPED} &gt; 2]?hang,1)
exten =&gt; s,6,SetVar(DIR-CONTEXT=default)
exten =&gt; s,7,DigitTimeout(3)	; Bell-Inbound
exten =&gt; s,8,ResponseTimeout(7)
exten =&gt; s,9,Background(custom/aa_1)	; Stealth Auto-Attendant
exten =&gt; t,1,Zapateller	; jump
exten =&gt; t,2,Wait(2)
exten =&gt; t,3,Zapateller
exten =&gt; t,4,Wait(1)
exten =&gt; t,5,Zapateller
exten =&gt; t,6,Hangup

[ext-group]
include =&gt; ext-group-custom
exten =&gt; 1,1,SetMusicOnHold(default)
exten =&gt; 1,2,Macro(rg-group,ringall,20,,250-260-280-290-261)
exten =&gt; 1,3,Macro(vm,250)	; jump

P.S.I have a incoming bell (pstn) line connected to a tdm card with 1 FXO and 1 FXS and sipura fxs device and some ip phones.]]></description>
			<content:encoded><![CDATA[<p>Fantastic article. Ever since I stepped into the voip telephony world, Ward and family have become my new best friends. Thank you for taking the time to write these articles, you are simply fantastic.</p>
<p>I have setup a nice Asterisk @home system, I got a stealth autoattendant going. I have implemented the ivr which says press 1 to continue which then gets forwarded to a ring group which rings all extensions and then gets dumped to an extensions vmail. If 1 is not received from the caller, then it is assumed this could be a telemarketer and a Zapateller sit tone is sent out, I am plagued with telemarketers!! My problem is, when the caller presses 1, i want them to hear music while all the phones ring, but i am not able to get the music on. Any ideas?</p>
<p>my extensions_additional.conf ivr code:</p>
<p>[aa_1]<br />
include => aa_1-custom<br />
exten => 1,1,Goto(ext-group,1,1)	; jump<br />
exten => fax,1,Goto(ext-fax,in_fax,1)<br />
exten => h,1,Hangup<br />
exten => hang,1,Playback(vm-goodbye)<br />
exten => hang,2,Hangup<br />
exten => i,1,Playback(invalid)<br />
exten => i,2,Goto(s,7)<br />
include => ext-local<br />
include => app-messagecenter<br />
include => app-directory<br />
exten => s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4)<br />
exten => s,2,Answer<br />
exten => s,3,Wait(1)<br />
exten => s,4,SetVar(LOOPED=1)<br />
exten => s,5,GotoIf($[${LOOPED} > 2]?hang,1)<br />
exten => s,6,SetVar(DIR-CONTEXT=default)<br />
exten => s,7,DigitTimeout(3)	; Bell-Inbound<br />
exten => s,8,ResponseTimeout(7)<br />
exten => s,9,Background(custom/aa_1)	; Stealth Auto-Attendant<br />
exten => t,1,Zapateller	; jump<br />
exten => t,2,Wait(2)<br />
exten => t,3,Zapateller<br />
exten => t,4,Wait(1)<br />
exten => t,5,Zapateller<br />
exten => t,6,Hangup</p>
<p>[ext-group]<br />
include => ext-group-custom<br />
exten => 1,1,SetMusicOnHold(default)<br />
exten => 1,2,Macro(rg-group,ringall,20,,250-260-280-290-261)<br />
exten => 1,3,Macro(vm,250)	; jump</p>
<p>P.S.I have a incoming bell (pstn) line connected to a tdm card with 1 FXO and 1 FXS and sipura fxs device and some ip phones.</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Uwe Burger		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1142</link>

		<dc:creator><![CDATA[Uwe Burger]]></dc:creator>
		<pubDate>Tue, 07 Mar 2006 15:39:35 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1142</guid>

					<description><![CDATA[Important Tip for Call-Forward-On-Busy

Insert the following Code into extensions.conf just above
; cancels call forward on busy for calling extension
exten =&gt; *91,1,Macro(user-callerid)

---%&lt; -------------------------------------------------
; activate Call-Forward-On-Busy (CFB) with *90
exten =&gt; *90,1,Answer
exten =&gt; *90,2,Wait(1)
exten =&gt; *90,3,BackGround(please-enter-your)
exten =&gt; *90,4,Playback(extension)
exten =&gt; *90,5,Read(fromext,then-press-pound)
exten =&gt; *90,6,Wait(1)
exten =&gt; *90,7,BackGround(ent-target-attendant)
exten =&gt; *90,8,Read(toext,then-press-pound)
exten =&gt; *90,9,Wait(1)
exten =&gt; *90,10,DBput(CFB/${fromext}=${toext})
exten =&gt; *90,11,Playback(call-fwd-on-busy)
exten =&gt; *90,12,Playback(for)
exten =&gt; *90,13,Playback(extension)
exten =&gt; *90,14,SayDigits(${fromext})
exten =&gt; *90,15,Playback(is-set-to)
exten =&gt; *90,16,SayDigits(${toext})
exten =&gt; *90,17,Macro(hangupcall)
-----%&lt;-------------------------------------------]]></description>
			<content:encoded><![CDATA[<p>Important Tip for Call-Forward-On-Busy</p>
<p>Insert the following Code into extensions.conf just above<br />
; cancels call forward on busy for calling extension<br />
exten => *91,1,Macro(user-callerid)</p>
<p>&#8212;%< -------------------------------------------------
; activate Call-Forward-On-Busy (CFB) with *90
exten => *90,1,Answer<br />
exten => *90,2,Wait(1)<br />
exten => *90,3,BackGround(please-enter-your)<br />
exten => *90,4,Playback(extension)<br />
exten => *90,5,Read(fromext,then-press-pound)<br />
exten => *90,6,Wait(1)<br />
exten => *90,7,BackGround(ent-target-attendant)<br />
exten => *90,8,Read(toext,then-press-pound)<br />
exten => *90,9,Wait(1)<br />
exten => *90,10,DBput(CFB/${fromext}=${toext})<br />
exten => *90,11,Playback(call-fwd-on-busy)<br />
exten => *90,12,Playback(for)<br />
exten => *90,13,Playback(extension)<br />
exten => *90,14,SayDigits(${fromext})<br />
exten => *90,15,Playback(is-set-to)<br />
exten => *90,16,SayDigits(${toext})<br />
exten => *90,17,Macro(hangupcall)<br />
&#8212;&#8211;%<-------------------------------------------
</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: TimC		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1131</link>

		<dc:creator><![CDATA[TimC]]></dc:creator>
		<pubDate>Mon, 06 Mar 2006 02:39:16 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1131</guid>

					<description><![CDATA[I know this isn&#039;t the Tech support site for Asterisk, but I have 1 question no one seems to be able to answer, Is there anything I can do to make call-waiting active on all extensions by default? It&#039;s driving me insane.

&lt;i&gt;[WM: The problem is that you have to set a flag to set call waiting OFF with Asterisk. Here&#039;s a &lt;a href=&quot;http://voxilla.com/index.php?name=PNphpBB2&amp;file=viewtopic&amp;t=7021&amp;highlight=call+waiting&amp;sid=8c89b6dd7d6335df9f4f46fb014cbbec&quot;&gt;thread on Voxilla&lt;/a&gt; that will show you how to do it quickly without picking up each phone.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>I know this isn&#8217;t the Tech support site for Asterisk, but I have 1 question no one seems to be able to answer, Is there anything I can do to make call-waiting active on all extensions by default? It&#8217;s driving me insane.</p>
<p><i>[WM: The problem is that you have to set a flag to set call waiting OFF with Asterisk. Here&#8217;s a <a href="http://voxilla.com/index.php?name=PNphpBB2&#038;file=viewtopic&#038;t=7021&#038;highlight=call+waiting&#038;sid=8c89b6dd7d6335df9f4f46fb014cbbec">thread on Voxilla</a> that will show you how to do it quickly without picking up each phone.]</i></p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Jim		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1129</link>

		<dc:creator><![CDATA[Jim]]></dc:creator>
		<pubDate>Sat, 04 Mar 2006 18:54:01 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1129</guid>

					<description><![CDATA[THANK YOU!!  Your articles and blogging are so helpful! I&#039;ve successfully setup my AAH and saved loads of time thanks to you. Keep up the great work!]]></description>
			<content:encoded><![CDATA[<p>THANK YOU!!  Your articles and blogging are so helpful! I&#8217;ve successfully setup my AAH and saved loads of time thanks to you. Keep up the great work!</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: RareSanity		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1124</link>

		<dc:creator><![CDATA[RareSanity]]></dc:creator>
		<pubDate>Fri, 03 Mar 2006 19:34:58 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1124</guid>

					<description><![CDATA[First of all, this article is top notch! Followed to the &#039;T&#039; and now have a system up and running. Two questions though, is it normal to have large delays in festival transacation (like the default weather)? Also, in testing with a softphone, and dialing a different extension, I get &quot;The person at extension&quot; sound, but no actual number, and then it just hangs up, anyone else seen this?

&lt;i&gt;[WM: Haven&#039;t seen the second problem you mentioned. But we have the same issue with long Festival delays which we didn&#039;t see at all in AAH 1.5. I&#039;ve mentioned to Andrew, but he doesn&#039;t see it. It may be a low memory, slow processor issue in my case. What type PC are you running?]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>First of all, this article is top notch! Followed to the &#8216;T&#8217; and now have a system up and running. Two questions though, is it normal to have large delays in festival transacation (like the default weather)? Also, in testing with a softphone, and dialing a different extension, I get "The person at extension" sound, but no actual number, and then it just hangs up, anyone else seen this?</p>
<p><i>[WM: Haven&#8217;t seen the second problem you mentioned. But we have the same issue with long Festival delays which we didn&#8217;t see at all in AAH 1.5. I&#8217;ve mentioned to Andrew, but he doesn&#8217;t see it. It may be a low memory, slow processor issue in my case. What type PC are you running?]</i></p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: AJ		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1099</link>

		<dc:creator><![CDATA[AJ]]></dc:creator>
		<pubDate>Mon, 27 Feb 2006 22:59:59 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1099</guid>

					<description><![CDATA[Anyone know how make different incoming trunks do different options?? It seems that no matter what config files I modify, ALL my incoming calls go to the incoming calls setting specified in AMP. Its as if something is overriding my DID incoming calls settings.  Anybody have an idea for me?

Thanks]]></description>
			<content:encoded><![CDATA[<p>Anyone know how make different incoming trunks do different options?? It seems that no matter what config files I modify, ALL my incoming calls go to the incoming calls setting specified in AMP. Its as if something is overriding my DID incoming calls settings.  Anybody have an idea for me?</p>
<p>Thanks</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Desmond		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1091</link>

		<dc:creator><![CDATA[Desmond]]></dc:creator>
		<pubDate>Sun, 26 Feb 2006 20:52:32 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1091</guid>

					<description><![CDATA[Hi,
Great articles!!!  I am still experiencing memory leak with Asterisk 2.5.  Is this a known issue or is it just me?

&lt;i&gt;[WM: Memory leak means things get worse and worse until your system ultimately crashes. Are you seeing crashes? Linux is designed to use all the memory it can get its hands on. So, just because your system shows less free resources on Day 2 than it did on Day 1 doesn&#039;t necessarily mean there&#039;s a memory leak. It may just be Linux caching more data to speed processing up. Tell us more.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Hi,<br />
Great articles!!!  I am still experiencing memory leak with Asterisk 2.5.  Is this a known issue or is it just me?</p>
<p><i>[WM: Memory leak means things get worse and worse until your system ultimately crashes. Are you seeing crashes? Linux is designed to use all the memory it can get its hands on. So, just because your system shows less free resources on Day 2 than it did on Day 1 doesn&#8217;t necessarily mean there&#8217;s a memory leak. It may just be Linux caching more data to speed processing up. Tell us more.]</i></p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: sean courtney		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1075</link>

		<dc:creator><![CDATA[sean courtney]]></dc:creator>
		<pubDate>Wed, 22 Feb 2006 19:05:43 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1075</guid>

					<description><![CDATA[i have asterisk 2.4, vonage analog lines, and vonage soft phone lines. i having issues with incoming calls. it appears that dtmf is not working for incoming calls on both vonage analog and soft phone accounts. i have tested the pbx using ext 7777 and everything is fine. 
 
the vonage analog lines are connected using a digium card.  
 
i have tried just about everything i can think of. is there a trick in getting vonage to send dtmf signals cleanly to my pbx, or is this a know issue? 
 
has anyone out there run into the same issue and resolved it? everything would be perfect if this issue could be resolved. any help would be greatly appreciated. 
 
thanks, 
sean]]></description>
			<content:encoded><![CDATA[<p>i have asterisk 2.4, vonage analog lines, and vonage soft phone lines. i having issues with incoming calls. it appears that dtmf is not working for incoming calls on both vonage analog and soft phone accounts. i have tested the pbx using ext 7777 and everything is fine. </p>
<p>the vonage analog lines are connected using a digium card.  </p>
<p>i have tried just about everything i can think of. is there a trick in getting vonage to send dtmf signals cleanly to my pbx, or is this a know issue? </p>
<p>has anyone out there run into the same issue and resolved it? everything would be perfect if this issue could be resolved. any help would be greatly appreciated. </p>
<p>thanks,<br />
sean</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Reginald		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1046</link>

		<dc:creator><![CDATA[Reginald]]></dc:creator>
		<pubDate>Sun, 19 Feb 2006 11:23:52 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1046</guid>

					<description><![CDATA[You&#039;ve recommended using Phlink boxes in your Mac section and have  gone into great detail about Asterisk on controlling phone  communications in Nerd Vittles. I&#039;m wondering which would be better  for a person or small business.
 
 I&#039;m more into Macs than Windows/Linux, but am familiar enough with computers to have no problem either way in implementing such a  system. My first computer was an Apple II+ back in 1979 and I&#039;ve  worked at three Apple dealers and four schools/school boards in my  time doing tech support work. While I can fix most any old Mac (OS9 and under) I&#039;m still getting used to OSX and use the computers as  computers not toys.
 
 My business has three lines on a rotary setup, an individual phone line used by the company that is also indicated as being with an  associated company, a single dedicated fax line for receiving faxes  and a single line that we use for computer equipment to dial out  (Western Union terminal, POS terminal, etc.). The last line of the  three line rotary is also used for a second Western Union terminal  dial-out line. I am contemplating replacing the DOS box Western Union  terminals with either small XP boxes or Macs capable of running  Windows and having them connect via the internet, but need to assure 
 myself of the security of doing so.
 
 I&#039;ve been thinking of getting a VOIP line for cheaper long distance  capabilities, having a Skype line for incoming calls as well as calls  to other Skype people and possibly other VOIP lines if needed for  specific VOIP features that might be available inexpensively  depending on what our primary Canadian VOIP might offer.
 
 I&#039;d like to be able to build a system whereby callerID would identify  who was calling and be able to populate a screen saying who was  calling, potentially record the details of certain calls, provide IVR  capability for those who call here seeking information on what we do,  etc.
 
 I keep tossing between Asterisk and using Phlink or Phone Valet  running on Macs. Each has their good and bad points, costs and being  able to tie everything together to make it as seamless as possible.  And since Asterisk has an OSX port, is it possible to do a Mac with  Asterisk and Phlink/Phone Valet in combination to get the best of  both worlds? Phlink/Phone Valet for land lines and Asterisk for  VOIP/Skype/etc.? (since Asterisk on the Mac doesn&#039;t seem to support  the Digium cards to connect land lines, or am I missing something?)
 
 Thanks for your input or suggestions on where to go for further info.  Some days there are too many options to choose from and I am working  on too many other projects to do proper service to these decisions as  well.

&lt;i&gt;[WM: For what you&#039;re talking about, it&#039;s Asterisk hands-down. If you don&#039;t want to go the Linux route, you might want to hold off a few weeks and try the new &lt;a href=&quot;http://voipspeak.net/index.php?/content/view/60/2/&quot;&gt;SPA-9000&lt;/a&gt; which is a preconfigured Asterisk system minus voicemail. But frankly, based on your experience, it sounds like Asterisk on Linux would be a walk in the park for you.  Any $500 PC with an IDE drive works. If I were running my business on it, I&#039;d probably want a spare as well. Read our articles. It&#039;ll reduce the learning curve by about 99%. If you want to get something up and running on a Windows PC just to try, you can&#039;t beat the new &lt;a href=&quot;http://mundy.org/blog/index.php?p=116&quot;&gt;VMware solution&lt;/a&gt;. I wouldn&#039;t run my business on it, but it&#039;ll quickly give you a good idea of what Asterisk can do. Best of luck.]&lt;/i&gt;
]]></description>
			<content:encoded><![CDATA[<p>You&#8217;ve recommended using Phlink boxes in your Mac section and have  gone into great detail about Asterisk on controlling phone  communications in Nerd Vittles. I&#8217;m wondering which would be better  for a person or small business.</p>
<p> I&#8217;m more into Macs than Windows/Linux, but am familiar enough with computers to have no problem either way in implementing such a  system. My first computer was an Apple II+ back in 1979 and I&#8217;ve  worked at three Apple dealers and four schools/school boards in my  time doing tech support work. While I can fix most any old Mac (OS9 and under) I&#8217;m still getting used to OSX and use the computers as  computers not toys.</p>
<p> My business has three lines on a rotary setup, an individual phone line used by the company that is also indicated as being with an  associated company, a single dedicated fax line for receiving faxes  and a single line that we use for computer equipment to dial out  (Western Union terminal, POS terminal, etc.). The last line of the  three line rotary is also used for a second Western Union terminal  dial-out line. I am contemplating replacing the DOS box Western Union  terminals with either small XP boxes or Macs capable of running  Windows and having them connect via the internet, but need to assure<br />
 myself of the security of doing so.</p>
<p> I&#8217;ve been thinking of getting a VOIP line for cheaper long distance  capabilities, having a Skype line for incoming calls as well as calls  to other Skype people and possibly other VOIP lines if needed for  specific VOIP features that might be available inexpensively  depending on what our primary Canadian VOIP might offer.</p>
<p> I&#8217;d like to be able to build a system whereby callerID would identify  who was calling and be able to populate a screen saying who was  calling, potentially record the details of certain calls, provide IVR  capability for those who call here seeking information on what we do,  etc.</p>
<p> I keep tossing between Asterisk and using Phlink or Phone Valet  running on Macs. Each has their good and bad points, costs and being  able to tie everything together to make it as seamless as possible.  And since Asterisk has an OSX port, is it possible to do a Mac with  Asterisk and Phlink/Phone Valet in combination to get the best of  both worlds? Phlink/Phone Valet for land lines and Asterisk for  VOIP/Skype/etc.? (since Asterisk on the Mac doesn&#8217;t seem to support  the Digium cards to connect land lines, or am I missing something?)</p>
<p> Thanks for your input or suggestions on where to go for further info.  Some days there are too many options to choose from and I am working  on too many other projects to do proper service to these decisions as  well.</p>
<p><i>[WM: For what you&#8217;re talking about, it&#8217;s Asterisk hands-down. If you don&#8217;t want to go the Linux route, you might want to hold off a few weeks and try the new <a href="http://voipspeak.net/index.php?/content/view/60/2/">SPA-9000</a> which is a preconfigured Asterisk system minus voicemail. But frankly, based on your experience, it sounds like Asterisk on Linux would be a walk in the park for you.  Any $500 PC with an IDE drive works. If I were running my business on it, I&#8217;d probably want a spare as well. Read our articles. It&#8217;ll reduce the learning curve by about 99%. If you want to get something up and running on a Windows PC just to try, you can&#8217;t beat the new <a href="http://mundy.org/blog/index.php?p=116">VMware solution</a>. I wouldn&#8217;t run my business on it, but it&#8217;ll quickly give you a good idea of what Asterisk can do. Best of luck.]</i></p>
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		<title>
		By: Blake		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1036</link>

		<dc:creator><![CDATA[Blake]]></dc:creator>
		<pubDate>Sat, 18 Feb 2006 20:04:29 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1036</guid>

					<description><![CDATA[Ward, think you could request a couple more icons for the a2billing and a2customer links? I love the blog, great work as always! I&#039;m hoping to start working on the Asteridex and getting that setup in my environment.]]></description>
			<content:encoded><![CDATA[<p>Ward, think you could request a couple more icons for the a2billing and a2customer links? I love the blog, great work as always! I&#8217;m hoping to start working on the Asteridex and getting that setup in my environment.</p>
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		<item>
		<title>
		By: Kristopher E.J.		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1029</link>

		<dc:creator><![CDATA[Kristopher E.J.]]></dc:creator>
		<pubDate>Sat, 18 Feb 2006 02:27:06 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1029</guid>

					<description><![CDATA[With regards to the NVFaxDetect... THANKS!!  That was a very easy change to impliment and is working great with my IAX trunks.

There was one thing I did end up adding however, as I noticed it would not detect incoming faxes to DIDs (Or Incoming Routing as it&#039;s now refered to)

I edited the webpage that adds the DID&#039;s to asterisk&#039;s configs and implimented your changes there.  AMP now has the option to answer immidiately and wait for a given amount of time - so instead of having it answer() and wait(x), it now will answer(), Playtones(ring), and NVFaxDetect(x).  Here&#039;s how:

Edit the file: /var/www/html/admin/did.php

Find the following section:

if ($answer == &quot;1&quot;) {
$addarray[] = array(&#039;ext-did&#039;,$account,$i++,&#039;Answer&#039;,&#039;&#039;,&#039;&#039;,&#039;0&#039;);
$addarray[] = array(&#039;ext-did&#039;,$account,$i++,&#039;Wait&#039;,$wait,&#039;&#039;,&#039;0&#039;);
}

and change it to:

if ($answer == &quot;1&quot;) {
$addarray[] = array(&#039;ext-did&#039;,$account,$i++,&#039;Answer&#039;,&#039;&#039;,&#039;&#039;,&#039;0&#039;);
$addarray[] = array(&#039;ext-did&#039;,$account,$i++,&#039;Playtones&#039;,&#039;ring&#039;,&#039;&#039;,&#039;0&#039;);
$addarray[] = array(&#039;ext-did&#039;,$account,$i++,&#039;NVFaxDetect&#039;,$wait,&#039;&#039;,&#039;0&#039;);
}

Now not only can you have incoming DID&#039;s accept faxes, but also have them emailed to different email addresses based on the fax handling options you specify in the AMP Incoming Route screen.

Keep up the great work, and thanks for all the help you&#039;ve been!!

-Kris

&lt;i&gt;[WM: Great suggestion. Thank you.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>With regards to the NVFaxDetect&#8230; THANKS!!  That was a very easy change to impliment and is working great with my IAX trunks.</p>
<p>There was one thing I did end up adding however, as I noticed it would not detect incoming faxes to DIDs (Or Incoming Routing as it&#8217;s now refered to)</p>
<p>I edited the webpage that adds the DID&#8217;s to asterisk&#8217;s configs and implimented your changes there.  AMP now has the option to answer immidiately and wait for a given amount of time &#8211; so instead of having it answer() and wait(x), it now will answer(), Playtones(ring), and NVFaxDetect(x).  Here&#8217;s how:</p>
<p>Edit the file: /var/www/html/admin/did.php</p>
<p>Find the following section:</p>
<p>if ($answer == "1&#8243;) {<br />
$addarray[] = array(&#8216;ext-did&#8217;,$account,$i++,&#8217;Answer&#8217;,",",&#8217;0&#8242;);<br />
$addarray[] = array(&#8216;ext-did&#8217;,$account,$i++,&#8217;Wait&#8217;,$wait,",&#8217;0&#8242;);<br />
}</p>
<p>and change it to:</p>
<p>if ($answer == "1&#8243;) {<br />
$addarray[] = array(&#8216;ext-did&#8217;,$account,$i++,&#8217;Answer&#8217;,",",&#8217;0&#8242;);<br />
$addarray[] = array(&#8216;ext-did&#8217;,$account,$i++,&#8217;Playtones&#8217;,&#8217;ring&#8217;,",&#8217;0&#8242;);<br />
$addarray[] = array(&#8216;ext-did&#8217;,$account,$i++,&#8217;NVFaxDetect&#8217;,$wait,",&#8217;0&#8242;);<br />
}</p>
<p>Now not only can you have incoming DID&#8217;s accept faxes, but also have them emailed to different email addresses based on the fax handling options you specify in the AMP Incoming Route screen.</p>
<p>Keep up the great work, and thanks for all the help you&#8217;ve been!!</p>
<p>-Kris</p>
<p><i>[WM: Great suggestion. Thank you.]</i></p>
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		<title>
		By: My name is Daniyl		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1017</link>

		<dc:creator><![CDATA[My name is Daniyl]]></dc:creator>
		<pubDate>Thu, 16 Feb 2006 01:21:30 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1017</guid>

					<description><![CDATA[Now, why can&#039;t I have this article in a PDF? or maybe in a printable form? Without the comments...
This is GOLD man!
Oh, and I haven&#039;t installed A@H yet. AND... Is there a way that I can use my existing (home) phone(s) seeing that I do not have an IP, SIP phone yet?
Without you, I think I would be lost. Thanx
Daniyl]]></description>
			<content:encoded><![CDATA[<p>Now, why can&#8217;t I have this article in a PDF? or maybe in a printable form? Without the comments&#8230;<br />
This is GOLD man!<br />
Oh, and I haven&#8217;t installed A@H yet. AND&#8230; Is there a way that I can use my existing (home) phone(s) seeing that I do not have an IP, SIP phone yet?<br />
Without you, I think I would be lost. Thanx<br />
Daniyl</p>
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		<title>
		By: Mike		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1015</link>

		<dc:creator><![CDATA[Mike]]></dc:creator>
		<pubDate>Wed, 15 Feb 2006 13:51:04 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1015</guid>

					<description><![CDATA[Wow!  This is a great site.  I have been playing with Asterisk@home and am looking at installing it in our office.  Everything is working great but I can&#039;t get call recording to work.  I can set recording to always in the extention set up and everything is great.  What I can&#039;t do is on demand recording.  I have the AAH 2.5 installed and have added the &#039;w&#039; and &#039;W&#039; to dialing options and also added [globals] 
 DYNAMIC_FEATURES =&gt; automon to extentions.conf but no joy in Muddville.  Any help would be greatly appreciated.

&lt;i&gt;[WM: Have you added &lt;b&gt;automon =&#062; *1 &lt;/b&gt; to the featuremap context of features.conf?]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Wow!  This is a great site.  I have been playing with Asterisk@home and am looking at installing it in our office.  Everything is working great but I can&#8217;t get call recording to work.  I can set recording to always in the extention set up and everything is great.  What I can&#8217;t do is on demand recording.  I have the AAH 2.5 installed and have added the &#8216;w&#8217; and &#8216;W&#8217; to dialing options and also added [globals]<br />
 DYNAMIC_FEATURES => automon to extentions.conf but no joy in Muddville.  Any help would be greatly appreciated.</p>
<p><i>[WM: Have you added <b>automon =&gt; *1 </b> to the featuremap context of features.conf?]</i></p>
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			</item>
		<item>
		<title>
		By: Gunner		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1008</link>

		<dc:creator><![CDATA[Gunner]]></dc:creator>
		<pubDate>Tue, 14 Feb 2006 16:14:26 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1008</guid>

					<description><![CDATA[It looks like I was wrong, btw... on the previous comment. I missed the close bracket after the !=. I&#039;m having trouble decifering what is going on with that line. 

A workaround... I removed that line and the line after it. It seems to work fine. 

If someone can help me figure out what&#039;s going on there, i&#039;d be very appriciative.]]></description>
			<content:encoded><![CDATA[<p>It looks like I was wrong, btw&#8230; on the previous comment. I missed the close bracket after the !=. I&#8217;m having trouble decifering what is going on with that line. </p>
<p>A workaround&#8230; I removed that line and the line after it. It seems to work fine. </p>
<p>If someone can help me figure out what&#8217;s going on there, i&#8217;d be very appriciative.</p>
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		<item>
		<title>
		By: Gunner		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1004</link>

		<dc:creator><![CDATA[Gunner]]></dc:creator>
		<pubDate>Mon, 13 Feb 2006 23:31:35 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1004</guid>

					<description><![CDATA[Because I seem to be good at finding problems, I&#039;ll answer the question above me first. Parts of Extensions.conf does not get written to, if you tweak it, the tweaks change. If you mess up something that amp needed to be able to make it function (E.G. a reference to a variable set in extensions_additional.conf) then it will be broken and you will have to replace the file or find where you broke it and fix it. Extensions_additional.conf is pretty much stored entierly in a mysql database, and whenever you change things through amp, it updates the database and rewrites the file. Extensions_custom.conf is for you to play with, and AMP doesn&#039;t need anything it it to make its system funciton. Other file sets have a simmilar pattern going on.

On to the problems... I was checking our logs today and noticed another syntax error being called out by the system (we had to reboot because of a memory leak, btw, and we&#039;ll see if this somehow had anything to do with it) but I traced it out and found this line in..

    [macro-rg-group]
    ...
    exten =&gt; s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} != ${RGPREFIX}]?4:3)  ; check for old prefix 

And I also had to trace down the meaning of TOK_NE (it&#039;s not equal token, btw) and figured out that there is a missing close bracket in the expression jut before the !=. So the line SHOULD look like this...
    
    exten =&gt; s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}}] != ${RGPREFIX}]?4:3)  ; check for old prefix

Thanks, Great job with the site btw. And regarding someone&#039;s table of requirements, The idea poped into my head of developing a web site that asks a series of questions about the requirements on thier side and it will return a sheet of requirements for a server to match... I might start work on that after tomorrow even.]]></description>
			<content:encoded><![CDATA[<p>Because I seem to be good at finding problems, I&#8217;ll answer the question above me first. Parts of Extensions.conf does not get written to, if you tweak it, the tweaks change. If you mess up something that amp needed to be able to make it function (E.G. a reference to a variable set in extensions_additional.conf) then it will be broken and you will have to replace the file or find where you broke it and fix it. Extensions_additional.conf is pretty much stored entierly in a mysql database, and whenever you change things through amp, it updates the database and rewrites the file. Extensions_custom.conf is for you to play with, and AMP doesn&#8217;t need anything it it to make its system funciton. Other file sets have a simmilar pattern going on.</p>
<p>On to the problems&#8230; I was checking our logs today and noticed another syntax error being called out by the system (we had to reboot because of a memory leak, btw, and we&#8217;ll see if this somehow had anything to do with it) but I traced it out and found this line in..</p>
<p>    [macro-rg-group]<br />
    &#8230;<br />
    exten => s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} != ${RGPREFIX}]?4:3)  ; check for old prefix </p>
<p>And I also had to trace down the meaning of TOK_NE (it&#8217;s not equal token, btw) and figured out that there is a missing close bracket in the expression jut before the !=. So the line SHOULD look like this&#8230;</p>
<p>    exten => s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}}] != ${RGPREFIX}]?4:3)  ; check for old prefix</p>
<p>Thanks, Great job with the site btw. And regarding someone&#8217;s table of requirements, The idea poped into my head of developing a web site that asks a series of questions about the requirements on thier side and it will return a sheet of requirements for a server to match&#8230; I might start work on that after tomorrow even.</p>
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		<item>
		<title>
		By: Anon		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-1000</link>

		<dc:creator><![CDATA[Anon]]></dc:creator>
		<pubDate>Mon, 13 Feb 2006 17:43:25 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-1000</guid>

					<description><![CDATA[I just finished setting up Asterisk (one afternoon, wao! the Linux learning curve dropped to zero). I am a newcomer and these are my questions: 

1) Why can&#039;t everything be managed from AMP?
2) I guess the opposite to AMP is to manually tweak the config files. In such case which one has precedence?
3) Am I totally lost?


Thank you for your answers and keep up the good work!]]></description>
			<content:encoded><![CDATA[<p>I just finished setting up Asterisk (one afternoon, wao! the Linux learning curve dropped to zero). I am a newcomer and these are my questions: </p>
<p>1) Why can&#8217;t everything be managed from AMP?<br />
2) I guess the opposite to AMP is to manually tweak the config files. In such case which one has precedence?<br />
3) Am I totally lost?</p>
<p>Thank you for your answers and keep up the good work!</p>
]]></content:encoded>
		
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		<item>
		<title>
		By: Kerry		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-999</link>

		<dc:creator><![CDATA[Kerry]]></dc:creator>
		<pubDate>Mon, 13 Feb 2006 16:36:05 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-999</guid>

					<description><![CDATA[Ward, any luck with phone paging in 2.5? I have tried it with 3 different installs and on all 3 systems the phones just ring and never pick up. Worked fine in AAH 2.1. The soundcard paging that you mentioned works fine (assuming you know how to turn the sound up on your sound card).]]></description>
			<content:encoded><![CDATA[<p>Ward, any luck with phone paging in 2.5? I have tried it with 3 different installs and on all 3 systems the phones just ring and never pick up. Worked fine in AAH 2.1. The soundcard paging that you mentioned works fine (assuming you know how to turn the sound up on your sound card).</p>
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			</item>
		<item>
		<title>
		By: Joni		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-996</link>

		<dc:creator><![CDATA[Joni]]></dc:creator>
		<pubDate>Mon, 13 Feb 2006 05:57:27 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-996</guid>

					<description><![CDATA[My AAH install is in jeopardy! My fiance just tried to call her parents house (local to us though I have all outgoing calls going through Telasip instead of the PSTN line) and it rang 3 or so times and then the phone went dead - no dial tone, no busy signal, nothing! I dialed my cell phone (also local to us) just to test and it went through just fine. We then tried her parent&#039;s number again and we got the same dead phone several more times before I unplugged the Sipura3000 from the PSTN and gave her a &quot;real&quot; phone. I haven&#039;t done anything to disable calls to her parents (I swear!) and am at a loss as it what&#039;s going on. I&#039;m no expert but nothing looked out of the ordinary in the logs and my install is pretty much a stock 2.5 install from this site with Telasip and a Sipura3000. If I can&#039;t resolve this I&#039;m sure my fiance will force me to 86 my newly installed AAH setup - help!

&lt;i&gt;[WM: Check the logs and turn on debugging in the CLI to see what&#039;s going on. This isn&#039;t a good forum for solving problems like this. Head on over to &lt;a href=&quot;http://voxilla.com/PNphpBB2-viewforum-f-17-sid-83a49889a62bdd10d6de80dd344743ef.html&quot;&gt;Voxilla&lt;/a&gt; and post your question once you have some better detail about the problem.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>My AAH install is in jeopardy! My fiance just tried to call her parents house (local to us though I have all outgoing calls going through Telasip instead of the PSTN line) and it rang 3 or so times and then the phone went dead &#8211; no dial tone, no busy signal, nothing! I dialed my cell phone (also local to us) just to test and it went through just fine. We then tried her parent&#8217;s number again and we got the same dead phone several more times before I unplugged the Sipura3000 from the PSTN and gave her a "real" phone. I haven&#8217;t done anything to disable calls to her parents (I swear!) and am at a loss as it what&#8217;s going on. I&#8217;m no expert but nothing looked out of the ordinary in the logs and my install is pretty much a stock 2.5 install from this site with Telasip and a Sipura3000. If I can&#8217;t resolve this I&#8217;m sure my fiance will force me to 86 my newly installed AAH setup &#8211; help!</p>
<p><i>[WM: Check the logs and turn on debugging in the CLI to see what&#8217;s going on. This isn&#8217;t a good forum for solving problems like this. Head on over to <a href="http://voxilla.com/PNphpBB2-viewforum-f-17-sid-83a49889a62bdd10d6de80dd344743ef.html">Voxilla</a> and post your question once you have some better detail about the problem.]</i></p>
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		<item>
		<title>
		By: Rob		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-995</link>

		<dc:creator><![CDATA[Rob]]></dc:creator>
		<pubDate>Mon, 13 Feb 2006 02:01:19 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-995</guid>

					<description><![CDATA[Are Yapper and Reminder fixed for 2.5?

&lt;i&gt;[WM: Not yet. TeleYapper 2.5 will be released on Tuesday, Feb. 14. The Telephone Reminder System still is in the hands of the programming department, moi.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Are Yapper and Reminder fixed for 2.5?</p>
<p><i>[WM: Not yet. TeleYapper 2.5 will be released on Tuesday, Feb. 14. The Telephone Reminder System still is in the hands of the programming department, moi.]</i></p>
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		<title>
		By: Elijah		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-991</link>

		<dc:creator><![CDATA[Elijah]]></dc:creator>
		<pubDate>Sun, 12 Feb 2006 12:49:05 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-991</guid>

					<description><![CDATA[I have enjoyed your articles. Let me ask one intriguing question. Can one use Asterisk for a community phone project wired and or wireless for about 5000 people to give them a cheap way to talk? I will appreciate your comments.

&lt;i&gt;[WM: Absolutely. Many are doing just that.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>I have enjoyed your articles. Let me ask one intriguing question. Can one use Asterisk for a community phone project wired and or wireless for about 5000 people to give them a cheap way to talk? I will appreciate your comments.</p>
<p><i>[WM: Absolutely. Many are doing just that.]</i></p>
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		<title>
		By: charles		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-988</link>

		<dc:creator><![CDATA[charles]]></dc:creator>
		<pubDate>Sat, 11 Feb 2006 17:19:42 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-988</guid>

					<description><![CDATA[We have an internal company portal where a new user can be added. User information is kept in a mysql database. We also want to use that portal to add asterisk extensions for a user. I would add couple lines in my portal application to write the extension to the mysql on asterisk server after I write into the portal mysql. So I don&#039;t have to add a new user through my portal and use AMP to create the extension. I am using Asterisk@Home 2.5, and we are using IAX clients.]]></description>
			<content:encoded><![CDATA[<p>We have an internal company portal where a new user can be added. User information is kept in a mysql database. We also want to use that portal to add asterisk extensions for a user. I would add couple lines in my portal application to write the extension to the mysql on asterisk server after I write into the portal mysql. So I don&#8217;t have to add a new user through my portal and use AMP to create the extension. I am using Asterisk@Home 2.5, and we are using IAX clients.</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: charles		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asterisk-at-home-25-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-986</link>

		<dc:creator><![CDATA[charles]]></dc:creator>
		<pubDate>Sat, 11 Feb 2006 15:08:19 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=112#comment-986</guid>

					<description><![CDATA[Great tutorial, love your site. I wonder if you know of a good tutorial on configuring asterisk realtime, for example add/deleting iax extensions, etc. The information on the web is always pieces here, pieces there. There is never a complete write up or example in one place. Thanks.

&lt;i&gt;[WM: Give me a little better hint, and we&#039;ll work on it.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Great tutorial, love your site. I wonder if you know of a good tutorial on configuring asterisk realtime, for example add/deleting iax extensions, etc. The information on the web is always pieces here, pieces there. There is never a complete write up or example in one place. Thanks.</p>
<p><i>[WM: Give me a little better hint, and we&#8217;ll work on it.]</i></p>
]]></content:encoded>
		
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