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	<title>
	Comments on: Newbie&#8217;s Guide to Asterisk@Home 2.4: Unabridged Soup-to-Nuts Installation Guide	</title>
	<atom:link href="https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/feed/" rel="self" type="application/rss+xml" />
	<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/</link>
	<description>Ward Mundy&#039;s Technobabblelog</description>
	<lastBuildDate>Thu, 02 Jun 2011 14:14:13 +0000</lastBuildDate>
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	<item>
		<title>
		By: Simon		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-933</link>

		<dc:creator><![CDATA[Simon]]></dc:creator>
		<pubDate>Fri, 03 Feb 2006 18:12:07 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-933</guid>

					<description><![CDATA[ever since 2.1 Centos installs but then asterisk refuses to compile and I&#039;m left with an asterisk1# prompt and nothing else.  I&#039;ve tried on an old compaq desktop and a recent dell server and I get the same result.  Any help would be much appreciated.  thanks - Simon

&lt;i&gt;[WM: If it&#039;s not a SATA drive, then I&#039;d suspect a bad ISO image or bad CD copy from the image.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>ever since 2.1 Centos installs but then asterisk refuses to compile and I&#8217;m left with an asterisk1# prompt and nothing else.  I&#8217;ve tried on an old compaq desktop and a recent dell server and I get the same result.  Any help would be much appreciated.  thanks &#8211; Simon</p>
<p><i>[WM: If it&#8217;s not a SATA drive, then I&#8217;d suspect a bad ISO image or bad CD copy from the image.]</i></p>
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		<title>
		By: Nilesh Londhe		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-926</link>

		<dc:creator><![CDATA[Nilesh Londhe]]></dc:creator>
		<pubDate>Fri, 03 Feb 2006 04:37:18 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-926</guid>

					<description><![CDATA[After finishing &quot;The Bleeding Edge&quot; instructions, when I start amportal again...here is what happens.

[root@asterisk1 src]# amportal start

SETTING FILE PERMISSIONS
chown: cannot access `/dev/dsp&#039;: No such file or directory
Permissions OK

STARTING ASTERISK
Asterisk ended with exit status 1
Asterisk died with code 1.
Automatically restarting Asterisk.

-----------------------------------------------------
Asterisk could not start!
Use &#039;tail /var/log/asterisk/full&#039; to find out why.
-----------------------------------------------------
[root@asterisk1 src]# tail /var/log/asterisk//full
Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Parsing &#039;/etc/asterisk/codecs.conf&#039;: Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Parsing &#039;/etc/asterisk/codecs.conf&#039;: Found
Feb  2 23:34:18 VERBOSE[16810] logger.c:     -- codec_gsm: using generic PLC
Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Registered translator &#039;gsmtolin&#039; from format gsm to slin, cost 4
Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Registered translator &#039;lintogsm&#039; from format slin to gsm, cost 9
Feb  2 23:34:18 VERBOSE[16810] logger.c:  [app_senddtmf.so]Feb  2 23:34:18 VERBOSE[16810] logger.c:  [app_senddtmf.so] =&gt; (Send DTMF digits Application)
Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Registered application &#039;SendDTMF&#039;
Feb  2 23:34:18 VERBOSE[16810] logger.c:  [format_vox.so]Feb  2 23:34:18 VERBOSE[16810] logger.c:  [format_vox.so] =&gt; (Dialogic VOX (ADPCM) File Format)
Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Registered file format vox, extension(s) vox
Feb  2 23:34:18 VERBOSE[16810] logger.c:  [app_txtcidname.so]Feb  2 23:34:18 WARNING[16810] loader.c: /usr/lib/asterisk/modules/app_txtcidname.so: undefined symbol: option_priority_jumping
Feb  2 23:34:18 WARNING[16810] loader.c: Loading module app_txtcidname.so failed!

Do you know why is asterisk dying?

&lt;i&gt;[WM: I do know why it&#039;s dying (the callerid module won&#039;t load), but I don&#039;t know what caused it. Save yourself a lot of headaches and install Asterisk@Home 2.5 which now includes Asterisk 1.2.4. Here&#039;s our &lt;a href=&quot;http://mundy.org/blog/index.php?p=112&quot;&gt;new tutorial&lt;/a&gt;.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>After finishing "The Bleeding Edge" instructions, when I start amportal again&#8230;here is what happens.</p>
<p>[root@asterisk1 src]# amportal start</p>
<p>SETTING FILE PERMISSIONS<br />
chown: cannot access `/dev/dsp&#8217;: No such file or directory<br />
Permissions OK</p>
<p>STARTING ASTERISK<br />
Asterisk ended with exit status 1<br />
Asterisk died with code 1.<br />
Automatically restarting Asterisk.</p>
<p>&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8211;<br />
Asterisk could not start!<br />
Use &#8216;tail /var/log/asterisk/full&#8217; to find out why.<br />
&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8212;&#8211;<br />
[root@asterisk1 src]# tail /var/log/asterisk//full<br />
Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Parsing &#8216;/etc/asterisk/codecs.conf&#8217;: Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Parsing &#8216;/etc/asterisk/codecs.conf&#8217;: Found<br />
Feb  2 23:34:18 VERBOSE[16810] logger.c:     &#8212; codec_gsm: using generic PLC<br />
Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Registered translator &#8216;gsmtolin&#8217; from format gsm to slin, cost 4<br />
Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Registered translator &#8216;lintogsm&#8217; from format slin to gsm, cost 9<br />
Feb  2 23:34:18 VERBOSE[16810] logger.c:  [app_senddtmf.so]Feb  2 23:34:18 VERBOSE[16810] logger.c:  [app_senddtmf.so] => (Send DTMF digits Application)<br />
Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Registered application &#8216;SendDTMF&#8217;<br />
Feb  2 23:34:18 VERBOSE[16810] logger.c:  [format_vox.so]Feb  2 23:34:18 VERBOSE[16810] logger.c:  [format_vox.so] => (Dialogic VOX (ADPCM) File Format)<br />
Feb  2 23:34:18 VERBOSE[16810] logger.c:   == Registered file format vox, extension(s) vox<br />
Feb  2 23:34:18 VERBOSE[16810] logger.c:  [app_txtcidname.so]Feb  2 23:34:18 WARNING[16810] loader.c: /usr/lib/asterisk/modules/app_txtcidname.so: undefined symbol: option_priority_jumping<br />
Feb  2 23:34:18 WARNING[16810] loader.c: Loading module app_txtcidname.so failed!</p>
<p>Do you know why is asterisk dying?</p>
<p><i>[WM: I do know why it&#8217;s dying (the callerid module won&#8217;t load), but I don&#8217;t know what caused it. Save yourself a lot of headaches and install Asterisk@Home 2.5 which now includes Asterisk 1.2.4. Here&#8217;s our <a href="http://mundy.org/blog/index.php?p=112">new tutorial</a>.]</i></p>
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		<title>
		By: JPE		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-925</link>

		<dc:creator><![CDATA[JPE]]></dc:creator>
		<pubDate>Fri, 03 Feb 2006 02:11:40 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-925</guid>

					<description><![CDATA[In response to the bluetooth. I got it working part ways going both directions. I can call out via the phone no problem. It will answer but gets stuck when I try to ring an extension. I am a very novice aah cat, so I know somebody who knows better can get it to work....
]]></description>
			<content:encoded><![CDATA[<p>In response to the bluetooth. I got it working part ways going both directions. I can call out via the phone no problem. It will answer but gets stuck when I try to ring an extension. I am a very novice aah cat, so I know somebody who knows better can get it to work&#8230;.</p>
]]></content:encoded>
		
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		<item>
		<title>
		By: Eric VDB		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-924</link>

		<dc:creator><![CDATA[Eric VDB]]></dc:creator>
		<pubDate>Thu, 02 Feb 2006 21:52:42 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-924</guid>

					<description><![CDATA[Hi Ward.

With great pleasure i read everytime your articles. Would it be possible to see a full install of all the individual components? (install OS, asterisk, amp, ...)

Also, maybe a tutorial on how to create trunk fallback would be cool. (ex: outbound call goes to trunk1, if thats not available go to trunk2, ....)

Or what about a more sophisticated dial plan. (Use provider A for calls to landlines, use provider B for calls to Belgian mobiles, use provider C for calls to Italian mobiles, etc)

Shame on me, so much questions...

Again, thanks for the great tutorials.

Eric]]></description>
			<content:encoded><![CDATA[<p>Hi Ward.</p>
<p>With great pleasure i read everytime your articles. Would it be possible to see a full install of all the individual components? (install OS, asterisk, amp, &#8230;)</p>
<p>Also, maybe a tutorial on how to create trunk fallback would be cool. (ex: outbound call goes to trunk1, if thats not available go to trunk2, &#8230;.)</p>
<p>Or what about a more sophisticated dial plan. (Use provider A for calls to landlines, use provider B for calls to Belgian mobiles, use provider C for calls to Italian mobiles, etc)</p>
<p>Shame on me, so much questions&#8230;</p>
<p>Again, thanks for the great tutorials.</p>
<p>Eric</p>
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		<title>
		By: Kaleb Kunzler		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-923</link>

		<dc:creator><![CDATA[Kaleb Kunzler]]></dc:creator>
		<pubDate>Thu, 02 Feb 2006 18:57:45 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-923</guid>

					<description><![CDATA[2.5 has been released officially, just thought I&#039;d let you know.]]></description>
			<content:encoded><![CDATA[<p>2.5 has been released officially, just thought I&#8217;d let you know.</p>
]]></content:encoded>
		
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		<item>
		<title>
		By: Gunner		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-921</link>

		<dc:creator><![CDATA[Gunner]]></dc:creator>
		<pubDate>Thu, 02 Feb 2006 17:33:08 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-921</guid>

					<description><![CDATA[Sorry to be such a pain in the ass, but going through my logs, I found this line...

&lt;i&gt;Feb  2 08:33:13 NOTICE[4630] pbx.c: Error in extension logic (missing &#039;}&#039;)&lt;/i&gt;

it would seem you are missing a } in the fix for max channels, if I&#039;m right. I could be wrong though.

Thanks for you&#039;re help!

&lt;i&gt;[WM: You&#039;re not a pain at all. Thanks for catching it. I&#039;ve fixed it in the article.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Sorry to be such a pain in the ass, but going through my logs, I found this line&#8230;</p>
<p><i>Feb  2 08:33:13 NOTICE[4630] pbx.c: Error in extension logic (missing &#8216;}&#8217;)</i></p>
<p>it would seem you are missing a } in the fix for max channels, if I&#8217;m right. I could be wrong though.</p>
<p>Thanks for you&#8217;re help!</p>
<p><i>[WM: You&#8217;re not a pain at all. Thanks for catching it. I&#8217;ve fixed it in the article.]</i></p>
]]></content:encoded>
		
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		<item>
		<title>
		By: Jeff		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-920</link>

		<dc:creator><![CDATA[Jeff]]></dc:creator>
		<pubDate>Thu, 02 Feb 2006 17:06:20 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-920</guid>

					<description><![CDATA[You said you enter your root password into http://192.168.0.108:10000 when prompted.  That is a non-routable IP, but doesn&#039;t that seem insecure?  Shouldn&#039;t you use httpS://192.168.0.108:10000? I haven&#039;t tried to install asterisk@home yet but I am about to deploy a bunch of Asterisk systems in production environments. (hints and consulting encouraged) ipstacks @no--spammgmail.com.

&lt;i&gt;[WM: Yes, it is insecure.  Unfortunately, https access isn&#039;t supported in the current default AAH configuration.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>You said you enter your root password into <a href="http://192.168.0.108:10000" rel="nofollow ugc">http://192.168.0.108:10000</a> when prompted.  That is a non-routable IP, but doesn&#8217;t that seem insecure?  Shouldn&#8217;t you use <a href="https://192.168.0.108:10000" rel="nofollow ugc">https://192.168.0.108:10000</a>? I haven&#8217;t tried to install asterisk@home yet but I am about to deploy a bunch of Asterisk systems in production environments. (hints and consulting encouraged) ipstacks @no&#8211;spammgmail.com.</p>
<p><i>[WM: Yes, it is insecure.  Unfortunately, https access isn&#8217;t supported in the current default AAH configuration.]</i></p>
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		<title>
		By: Thunderbird 1		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-918</link>

		<dc:creator><![CDATA[Thunderbird 1]]></dc:creator>
		<pubDate>Thu, 02 Feb 2006 13:00:55 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-918</guid>

					<description><![CDATA[Hi Ward!

Thank you for the &quot;thanks&quot; in &quot;Incoming Fax Support&quot; but in all honesty it is YOU that needs to be acknowledged for your previous article on installing &amp; configuring NVFaxDetect. I used your original faxing guide to configure my setup for incoming faxes &amp; during testing found a few anomalies with the priorities &amp; syntax which I rectified &amp; posted on the Whirlpool forum for others to benefit from.

The credit is rightfully yours :)

I get my satisfaction from the knowledge that the collaboration &amp; sharing of information helps others who may be less knowledgeable.

May I take this opportunity to commend you on all the various articles you have written so far, which I take great pleasure in reading &amp; look forward to your future articles.

Regards,
Thunderbird 1
***Share what you know - Learn what you don&#039;t***]]></description>
			<content:encoded><![CDATA[<p>Hi Ward!</p>
<p>Thank you for the "thanks" in "Incoming Fax Support" but in all honesty it is YOU that needs to be acknowledged for your previous article on installing &#038; configuring NVFaxDetect. I used your original faxing guide to configure my setup for incoming faxes &#038; during testing found a few anomalies with the priorities &#038; syntax which I rectified &#038; posted on the Whirlpool forum for others to benefit from.</p>
<p>The credit is rightfully yours ðŸ™‚</p>
<p>I get my satisfaction from the knowledge that the collaboration &#038; sharing of information helps others who may be less knowledgeable.</p>
<p>May I take this opportunity to commend you on all the various articles you have written so far, which I take great pleasure in reading &#038; look forward to your future articles.</p>
<p>Regards,<br />
Thunderbird 1<br />
***Share what you know &#8211; Learn what you don&#8217;t***</p>
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		<title>
		By: Nilesh Londhe		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-915</link>

		<dc:creator><![CDATA[Nilesh Londhe]]></dc:creator>
		<pubDate>Thu, 02 Feb 2006 06:08:30 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-915</guid>

					<description><![CDATA[Hi Ward: Thanks for a great series of articles on asterisk. BTW, with reference to comment number 7, instructions for using bluetooth as a trunk work only partially - only for dialling calls originating from asterisk and using Cellphone as trunk. They have not worked for me to receive calls into asterisk via bluetooth connected cellphone. Were you able to get it to work bidirectionally?

&lt;i&gt;[WM: I haven&#039;t messed with this. Bluetooth is so quirky to begin with on the Treo 650 that I doubt it would work in either direction. We need someone with a good Bluetooth phone to wrestle with it.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Hi Ward: Thanks for a great series of articles on asterisk. BTW, with reference to comment number 7, instructions for using bluetooth as a trunk work only partially &#8211; only for dialling calls originating from asterisk and using Cellphone as trunk. They have not worked for me to receive calls into asterisk via bluetooth connected cellphone. Were you able to get it to work bidirectionally?</p>
<p><i>[WM: I haven&#8217;t messed with this. Bluetooth is so quirky to begin with on the Treo 650 that I doubt it would work in either direction. We need someone with a good Bluetooth phone to wrestle with it.]</i></p>
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		<title>
		By: Michael J. Flood		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-914</link>

		<dc:creator><![CDATA[Michael J. Flood]]></dc:creator>
		<pubDate>Wed, 01 Feb 2006 23:05:23 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-914</guid>

					<description><![CDATA[I just completed &quot;The Bleeding Edge&quot; on a Compaq DL380 G1 with an exising A@H 2.4 configuration...Worked Perfectly!  Thanks for &quot;ALL&quot; of your hard work and I hope that you continue to provide these &quot;Excellent&quot; articles.

Thanks Again!]]></description>
			<content:encoded><![CDATA[<p>I just completed "The Bleeding Edge" on a Compaq DL380 G1 with an exising A@H 2.4 configuration&#8230;Worked Perfectly!  Thanks for "ALL" of your hard work and I hope that you continue to provide these "Excellent" articles.</p>
<p>Thanks Again!</p>
]]></content:encoded>
		
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		<title>
		By: Robert Moskowitz		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-913</link>

		<dc:creator><![CDATA[Robert Moskowitz]]></dc:creator>
		<pubDate>Wed, 01 Feb 2006 20:28:23 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-913</guid>

					<description><![CDATA[Comments:

Bluetooth support:

Does this mean I can use a bluetooth phone as a ZAP trunk?  Much neater, and cheaper than getting a &#039;Dock-N-Talk&#039; http://cgi.ebay.com/Dock-N-Talk-Bluetooth-Dial-Tone-From-Cellular_W0QQitemZ6472881835QQcategoryZ3311QQssPageNameZWD1VQQrdZ1QQcmdZViewItem

hopefully not a resource hog...

&lt;i&gt;[WM: As for Bluetooth as a ZAP trunk, take a look at &lt;a href=&quot;http ://www.thetechguide.com/howto/asterisk/chanbluetooth.html&quot;&gt;this article&lt;/a&gt;.]&lt;/i&gt;

GOT to get more memory!

You do not have a section on getting your Networking set up.  perhaps at least some discussion in the webmin, or is it assumed if you got this far you can do that?

&lt;i&gt;[WM: Good suggestion. I&#039;ve added it although we&#039;ve covered it in some of our previous articles. But it&#039;s good to get everything in one place so THANKS.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Comments:</p>
<p>Bluetooth support:</p>
<p>Does this mean I can use a bluetooth phone as a ZAP trunk?  Much neater, and cheaper than getting a &#8216;Dock-N-Talk&#8217; <a href="http://cgi.ebay.com/Dock-N-Talk-Bluetooth-Dial-Tone-From-Cellular_W0QQitemZ6472881835QQcategoryZ3311QQssPageNameZWD1VQQrdZ1QQcmdZViewItem" rel="nofollow ugc">http://cgi.ebay.com/Dock-N-Talk-Bluetooth-Dial-Tone-From-Cellular_W0QQitemZ6472881835QQcategoryZ3311QQssPageNameZWD1VQQrdZ1QQcmdZViewItem</a></p>
<p>hopefully not a resource hog&#8230;</p>
<p><i>[WM: As for Bluetooth as a ZAP trunk, take a look at <a href="http ://www.thetechguide.com/howto/asterisk/chanbluetooth.html">this article</a>.]</i></p>
<p>GOT to get more memory!</p>
<p>You do not have a section on getting your Networking set up.  perhaps at least some discussion in the webmin, or is it assumed if you got this far you can do that?</p>
<p><i>[WM: Good suggestion. I&#8217;ve added it although we&#8217;ve covered it in some of our previous articles. But it&#8217;s good to get everything in one place so THANKS.]</i></p>
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		<title>
		By: Gunner		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-912</link>

		<dc:creator><![CDATA[Gunner]]></dc:creator>
		<pubDate>Wed, 01 Feb 2006 19:26:56 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-912</guid>

					<description><![CDATA[So I&#039;m trying to get switched over to Asterisk@home using 2 quad span Digium cards for our old Proprietary PBX, and when the behavior switched from the ringall after hours behavior to the hunt group this morning (just put in in kind of hybrid style for testing last night) and noticed that we were not recieving any calls on the KSU (FXS&#039;s were being used so that the asterisk box had control, but the KSU users didn&#039;t notice too much). I loooked in the log file and noticed a syntax error durring the incomming calls that didn&#039;t ever ring. It said there was a missing &quot;]&quot; and so I set off tracing the dial plan. I&#039;ll be testing it tonight, but the default install of AAH 2.4 has a syntax error in extensions.conf under the macro &quot;macro-dial&quot; on line s,22
it needs to look like this, exten =&gt; s,22,GotoIf($[${HuntMembers} &gt;= 1]?30 , I think.

&lt;i&gt;[WM: You&#039;re right. There is a problem. There&#039;s one too many $[&#039;s after the GotoIf(. Here&#039;s what I think needs to happen, and I&#039;ll update the tutorial as well.]&lt;/i&gt;
&lt;u&gt;Before&lt;/u&gt;:
&lt;code&gt;
exten =&gt; s,22,GotoIf($[$[${HuntMembers} &gt;= 1]?30 )
&lt;/code&gt;
&lt;u&gt;After&lt;/u&gt;:
&lt;code&gt;
exten =&gt; s,22,GotoIf($[${HuntMembers} &gt;= 1]?30 )
&lt;/code&gt;
]]></description>
			<content:encoded><![CDATA[<p>So I&#8217;m trying to get switched over to Asterisk@home using 2 quad span Digium cards for our old Proprietary PBX, and when the behavior switched from the ringall after hours behavior to the hunt group this morning (just put in in kind of hybrid style for testing last night) and noticed that we were not recieving any calls on the KSU (FXS&#8217;s were being used so that the asterisk box had control, but the KSU users didn&#8217;t notice too much). I loooked in the log file and noticed a syntax error durring the incomming calls that didn&#8217;t ever ring. It said there was a missing "]" and so I set off tracing the dial plan. I&#8217;ll be testing it tonight, but the default install of AAH 2.4 has a syntax error in extensions.conf under the macro "macro-dial" on line s,22<br />
it needs to look like this, exten => s,22,GotoIf($[${HuntMembers} >= 1]?30 , I think.</p>
<p><i>[WM: You&#8217;re right. There is a problem. There&#8217;s one too many $[&#8216;s after the GotoIf(. Here&#8217;s what I think needs to happen, and I&#8217;ll update the tutorial as well.]</i><br />
<u>Before</u>:<br />
<code><br />
exten => s,22,GotoIf($[$[${HuntMembers} >= 1]?30 )<br />
</code><br />
<u>After</u>:<br />
<code><br />
exten => s,22,GotoIf($[${HuntMembers} >= 1]?30 )<br />
</code></p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Syd Carter		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-911</link>

		<dc:creator><![CDATA[Syd Carter]]></dc:creator>
		<pubDate>Wed, 01 Feb 2006 18:22:00 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-911</guid>

					<description><![CDATA[Hey Ward.. thanks for being so great at keeping me in the know about asterisk.  What can be done about those of us who dont want to kill off our existing config files and yet still upgrade?

&lt;i&gt;[WM: Great question. Same old answer, I&#039;m afraid.  Call up each page in AMP and print the screen either to PDF or to a printer. And then start retyping. Good luck.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Hey Ward.. thanks for being so great at keeping me in the know about asterisk.  What can be done about those of us who dont want to kill off our existing config files and yet still upgrade?</p>
<p><i>[WM: Great question. Same old answer, I&#8217;m afraid.  Call up each page in AMP and print the screen either to PDF or to a printer. And then start retyping. Good luck.]</i></p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Ryan Lusvardi		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-910</link>

		<dc:creator><![CDATA[Ryan Lusvardi]]></dc:creator>
		<pubDate>Wed, 01 Feb 2006 17:58:43 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-910</guid>

					<description><![CDATA[Ward, with out you I would be no where with Asterisk.  Thanks for your diligence.  

FYI it has been my experience that Call Recording files are saved here /var/spool/asterisk/monitor by default not /var/lib/asterisk/monitor.  

I would love to see a good tutorial on meetme in the future.  Thanks...

&lt;i&gt;[WM: You&#039;re right on the storage location of Call Recording files. Seems like I spend half my life in /var/lib/asterisk, and I just type it like a robot every time anything starts with var.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Ward, with out you I would be no where with Asterisk.  Thanks for your diligence.  </p>
<p>FYI it has been my experience that Call Recording files are saved here /var/spool/asterisk/monitor by default not /var/lib/asterisk/monitor.  </p>
<p>I would love to see a good tutorial on meetme in the future.  Thanks&#8230;</p>
<p><i>[WM: You&#8217;re right on the storage location of Call Recording files. Seems like I spend half my life in /var/lib/asterisk, and I just type it like a robot every time anything starts with var.]</i></p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Rob Adelman		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-909</link>

		<dc:creator><![CDATA[Rob Adelman]]></dc:creator>
		<pubDate>Wed, 01 Feb 2006 15:46:48 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-909</guid>

					<description><![CDATA[Ward - aheads up about 2.4, I haven&#039;t been able to get Teleyapper to work - it was perfect in 2.2.

&lt;i&gt;[WM: Yep. Both TeleYapper and the Telephone Reminder System are broken in 2.4. They both are going to need some retrofitting... hopefully next week.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Ward &#8211; aheads up about 2.4, I haven&#8217;t been able to get Teleyapper to work &#8211; it was perfect in 2.2.</p>
<p><i>[WM: Yep. Both TeleYapper and the Telephone Reminder System are broken in 2.4. They both are going to need some retrofitting&#8230; hopefully next week.]</i></p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: matt avila		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-908</link>

		<dc:creator><![CDATA[matt avila]]></dc:creator>
		<pubDate>Wed, 01 Feb 2006 13:42:27 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-908</guid>

					<description><![CDATA[]]></description>
			<content:encoded><![CDATA[<p>Ward &#8211; again, excellent articles! Keep &#8217;em coming. One "tid bit" Ive picked up in chatting with Gene at TelaSIP is the ability to change the outbound CID strings. Here is the scenario &#8211; Once you start working wiht Asterisk, I suspect that many folk will take advantage of Stanaphone, IPKall, and other "free" inbound SIP services (also may have a POTS line attached through an X100P card for fail safe 911 dialing). </p>
<p>Using a multi line SIP phone (I?¢ÀÜ?°?Ç¬¢?¢‚Ç¨?°?É‚Ä°?Ç¬®?¢‚Ç¨?°?É‚Äò?Ç¬¢ve got a few Grandstream GXP2000&#8217;s) one can create multiple accounts and dial the appropriate account &#8211; while setting the CID to appear that its from a freebie SIP line. Gene said its actually VERY simple for him &#8211; he clears the CID and YOU have to set it&#8230; simply put "sendrpid=yes" into your SIP configuration and BE SURE to set the CID for the appropriate trunk &#038;/or extension (I use the extension to override the trunk CID for spoofing SIP numbers). Its a neat trick and still lets you use ONE great service (TelaSIP) and still have many inbound trunks.</p>
<p><i>[WM: Neat trick! Can&#8217;t wait to try it.]</i></p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: DUsty		</title>
		<link>https://nerdvittles.com/newbies-guide-to-asteriskhome-24-unabridged-soup-to-nuts-installation-guide/comment-page-1/#comment-907</link>

		<dc:creator><![CDATA[DUsty]]></dc:creator>
		<pubDate>Wed, 01 Feb 2006 12:22:12 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=110#comment-907</guid>

					<description><![CDATA[I installed fresh from the ISO, and MWI is not working. Did you notice that with your setup. Also. I had to manually create the /var/spool/asterisk/voicemail directory to be able to save a VM. Otherwise when it went to voice mail it would not allow a caller to leave you a voicemail.

&lt;i&gt;[WM: Ours works fine. Did you configure voicemail in AMP for the extension you created? This automatically sets up the necessary directories to support voicemail.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>I installed fresh from the ISO, and MWI is not working. Did you notice that with your setup. Also. I had to manually create the /var/spool/asterisk/voicemail directory to be able to save a VM. Otherwise when it went to voice mail it would not allow a caller to leave you a voicemail.</p>
<p><i>[WM: Ours works fine. Did you configure voicemail in AMP for the extension you created? This automatically sets up the necessary directories to support voicemail.]</i></p>
]]></content:encoded>
		
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	</channel>
</rss>
