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	<title>
	Comments on: Incredible PBX for Asterisk 1.8: Back from the Brink	</title>
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	<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/</link>
	<description>Ward Mundy&#039;s Technobabblelog</description>
	<lastBuildDate>Tue, 08 Dec 2015 15:43:51 +0000</lastBuildDate>
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	<item>
		<title>
		By: Ian Worthington		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-14402</link>

		<dc:creator><![CDATA[Ian Worthington]]></dc:creator>
		<pubDate>Sun, 08 May 2011 16:44:31 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-14402</guid>

					<description><![CDATA[Looks like even passwd-master is not required on 17561.

&lt;i&gt;[WM: Right you are. New article will be out tomorrow. New IncrediblePBX.com web site is already &quot;LIVE&quot; :-) ]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Looks like even passwd-master is not required on 17561.</p>
<p><i>[WM: Right you are. New article will be out tomorrow. New IncrediblePBX.com web site is already "LIVE" 🙂 ]</i></p>
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		<item>
		<title>
		By: ward		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13865</link>

		<dc:creator><![CDATA[ward]]></dc:creator>
		<pubDate>Tue, 22 Mar 2011 21:13:32 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13865</guid>

					<description><![CDATA[A new Google Voice quirk appeared today. If you have problems receiving incoming calls, visit &lt;a href=&quot;http://nerd.bz/emCyWQ&quot; rel=&quot;nofollow&quot;&gt;this thread&lt;/a&gt; on the PIAF Forums for the fix.]]></description>
			<content:encoded><![CDATA[<p>A new Google Voice quirk appeared today. If you have problems receiving incoming calls, visit <a href="http://nerd.bz/emCyWQ" rel="nofollow">this thread</a> on the PIAF Forums for the fix.</p>
]]></content:encoded>
		
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		<item>
		<title>
		By: Bill Dengler		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13842</link>

		<dc:creator><![CDATA[Bill Dengler]]></dc:creator>
		<pubDate>Fri, 18 Mar 2011 23:19:43 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13842</guid>

					<description><![CDATA[Can you make this work on fedora?

Or do I have to roll my own GV system?

&lt;i&gt;[WM: Start here: http://pbxinaflash.com/forum/showthread.php?t=7648 ] &lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Can you make this work on fedora?</p>
<p>Or do I have to roll my own GV system?</p>
<p><i>[WM: Start here: <a href="http://pbxinaflash.com/forum/showthread.php?t=7648" rel="nofollow ugc">http://pbxinaflash.com/forum/showthread.php?t=7648</a> ] </i></p>
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		<title>
		By: Paul N		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13755</link>

		<dc:creator><![CDATA[Paul N]]></dc:creator>
		<pubDate>Mon, 07 Mar 2011 05:49:23 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13755</guid>

					<description><![CDATA[All setup and working on latest build, however I cant gt my Cisco SPA501G to register from outside the building, only when on the LAN. Apart from that I have GV in and out with full audio working great!]]></description>
			<content:encoded><![CDATA[<p>All setup and working on latest build, however I cant gt my Cisco SPA501G to register from outside the building, only when on the LAN. Apart from that I have GV in and out with full audio working great!</p>
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		<item>
		<title>
		By: Paul N		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13721</link>

		<dc:creator><![CDATA[Paul N]]></dc:creator>
		<pubDate>Tue, 01 Mar 2011 17:30:38 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13721</guid>

					<description><![CDATA[I was installing PIAF 1.7.5.5.4 w/purple. All worked fine, downloaded and patched everything from the Internet, but Asterisk wont start as it cannot find some folder (see &lt;a href=&quot;https://picasaweb.google.com/pauljnye/PIAFScreenshots?authkey=Gv1sRgCJ2Xxbn6_KLPAQ#
&quot;&gt;3 screen shots&lt;/a&gt;..) I tried this on 2 separate new computers.
Any ideas?


&lt;i&gt;[WM: Be sure the clock is set correctly on your server before you begin. This will cause the Asterisk compile to fail. If that doesn&#039;t fix it, then you&#039;ve got network or DNS problems that are causing the downloads to abort. Just installed a new version without any problems.]&lt;/i&gt; ]]></description>
			<content:encoded><![CDATA[<p>I was installing PIAF 1.7.5.5.4 w/purple. All worked fine, downloaded and patched everything from the Internet, but Asterisk wont start as it cannot find some folder (see <a href="https://picasaweb.google.com/pauljnye/PIAFScreenshots?authkey=Gv1sRgCJ2Xxbn6_KLPAQ#
">3 screen shots</a>..) I tried this on 2 separate new computers.<br />
Any ideas?</p>
<p><i>[WM: Be sure the clock is set correctly on your server before you begin. This will cause the Asterisk compile to fail. If that doesn&#8217;t fix it, then you&#8217;ve got network or DNS problems that are causing the downloads to abort. Just installed a new version without any problems.]</i> </p>
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		<title>
		By: Glenn Jensen		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13697</link>

		<dc:creator><![CDATA[Glenn Jensen]]></dc:creator>
		<pubDate>Sat, 26 Feb 2011 18:33:15 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13697</guid>

					<description><![CDATA[Guys... I am testing the latest version in a virtualbox vm - can I move my freepbx configs from on Centos 5.x from a backup) over into Incredible PBX?

Also - what VM engine do most of you use?

&lt;i&gt;[WM: #1: Probably not if you want to preserve the Incredible PBX functionality. #2: Proxmox or RentPBX.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Guys&#8230; I am testing the latest version in a virtualbox vm &#8211; can I move my freepbx configs from on Centos 5.x from a backup) over into Incredible PBX?</p>
<p>Also &#8211; what VM engine do most of you use?</p>
<p><i>[WM: #1: Probably not if you want to preserve the Incredible PBX functionality. #2: Proxmox or RentPBX.]</i></p>
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		<title>
		By: Meech		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13630</link>

		<dc:creator><![CDATA[Meech]]></dc:creator>
		<pubDate>Tue, 15 Feb 2011 20:34:01 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13630</guid>

					<description><![CDATA[Hi there. First and foremost, thank you for all your work!! This is amazing.

During the asterisk compile process i&#039;m getting these three error lines:
iCBSearch.c: In function ‘iCBSearch’:
iCBSearch.c:130: warning: operation on ‘pp’ may be undefined
iCBSearch.c:325: warning: operation on ‘pp’ may be undefined

I found some talk about this error from a few years ago via the google machine, but i&#039;m not sure what to make of it or if it&#039;s OK to ignore.

The only place i found with a suggestion is here: https://issues.asterisk.org/view.php?id=12364

where they said to uncheck ilbc in the menuselect format list...

Let me know if there&#039;s any additional info i can provide. And i should note that i&#039;m new to the linux/sip scene so be gentle :)

Thanks!!!

&lt;i&gt;[WM: Head to the &lt;a href=&quot;http://pbxinaflash.com/forum/&quot; rel=&quot;nofollow&quot;&gt;PIAF Forums&lt;/a&gt;. It&#039;s too hard to diagnose issues like this on the blog.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Hi there. First and foremost, thank you for all your work!! This is amazing.</p>
<p>During the asterisk compile process i&#8217;m getting these three error lines:<br />
iCBSearch.c: In function ‘iCBSearch’:<br />
iCBSearch.c:130: warning: operation on ‘pp’ may be undefined<br />
iCBSearch.c:325: warning: operation on ‘pp’ may be undefined</p>
<p>I found some talk about this error from a few years ago via the google machine, but i&#8217;m not sure what to make of it or if it&#8217;s OK to ignore.</p>
<p>The only place i found with a suggestion is here: <a href="https://issues.asterisk.org/view.php?id=12364" rel="nofollow ugc">https://issues.asterisk.org/view.php?id=12364</a></p>
<p>where they said to uncheck ilbc in the menuselect format list&#8230;</p>
<p>Let me know if there&#8217;s any additional info i can provide. And i should note that i&#8217;m new to the linux/sip scene so be gentle 🙂</p>
<p>Thanks!!!</p>
<p><i>[WM: Head to the <a href="http://pbxinaflash.com/forum/" rel="nofollow">PIAF Forums</a>. It&#8217;s too hard to diagnose issues like this on the blog.]</i></p>
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		<title>
		By: ccg121		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13536</link>

		<dc:creator><![CDATA[ccg121]]></dc:creator>
		<pubDate>Tue, 01 Feb 2011 07:24:42 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13536</guid>

					<description><![CDATA[is there a reason their is no longer an a2billing install script?

&lt;i&gt;[WM: It now uses a version of PHP that is not supported by either CentOS or PBX in a Flash.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>is there a reason their is no longer an a2billing install script?</p>
<p><i>[WM: It now uses a version of PHP that is not supported by either CentOS or PBX in a Flash.]</i></p>
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		<title>
		By: thund3rhawk		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13535</link>

		<dc:creator><![CDATA[thund3rhawk]]></dc:creator>
		<pubDate>Mon, 31 Jan 2011 19:20:22 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13535</guid>

					<description><![CDATA[&lt;blockquote&gt;I got this working, and outbound works, but for incoming calls, my SIP station doesn’t get a call until the 4th ring to my dedicated GV number, and I only get MAYBE one ring before GV mail answers. I am using the above recommendation for allowing GV mail to work.
&lt;/blockquote&gt;


I have this same problem, and I guess it&#039;s because Google Voice only allows so many rings before it kicks in it&#039;s answering service. 
 
Does that sound right?

&lt;i&gt;[WM: Yep. That&#039;s the drawback at the moment since you can&#039;t alter the number of rings before Google&#039;s voicemail kicks in.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<blockquote><p>I got this working, and outbound works, but for incoming calls, my SIP station doesn’t get a call until the 4th ring to my dedicated GV number, and I only get MAYBE one ring before GV mail answers. I am using the above recommendation for allowing GV mail to work.
</p></blockquote>
<p>I have this same problem, and I guess it&#8217;s because Google Voice only allows so many rings before it kicks in it&#8217;s answering service. </p>
<p>Does that sound right?</p>
<p><i>[WM: Yep. That&#8217;s the drawback at the moment since you can&#8217;t alter the number of rings before Google&#8217;s voicemail kicks in.]</i></p>
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		<title>
		By: Gregory Gleason		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13518</link>

		<dc:creator><![CDATA[Gregory Gleason]]></dc:creator>
		<pubDate>Thu, 27 Jan 2011 15:47:27 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13518</guid>

					<description><![CDATA[I got this working, and outbound works, but for incoming calls, my SIP station doesn&#039;t get a call until the 4th ring to my dedicated GV number, and I only get MAYBE one ring before GV mail answers.  I am using the above recommendation for allowing GV mail to work.]]></description>
			<content:encoded><![CDATA[<p>I got this working, and outbound works, but for incoming calls, my SIP station doesn&#8217;t get a call until the 4th ring to my dedicated GV number, and I only get MAYBE one ring before GV mail answers.  I am using the above recommendation for allowing GV mail to work.</p>
]]></content:encoded>
		
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		<title>
		By: kevin		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13495</link>

		<dc:creator><![CDATA[kevin]]></dc:creator>
		<pubDate>Sat, 22 Jan 2011 21:05:50 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13495</guid>

					<description><![CDATA[I followed these directions and everything seems fine except I have no 701 extension.  Do I need to add that manually?

&lt;i&gt;[WM: If there&#039;s no 701 extension then some download failed during the install. Start over for best results.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>I followed these directions and everything seems fine except I have no 701 extension.  Do I need to add that manually?</p>
<p><i>[WM: If there&#8217;s no 701 extension then some download failed during the install. Start over for best results.]</i></p>
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		<title>
		By: ward		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13464</link>

		<dc:creator><![CDATA[ward]]></dc:creator>
		<pubDate>Tue, 18 Jan 2011 22:01:45 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13464</guid>

					<description><![CDATA[A major SIP security vulnerability was discovered in all versions of Asterisk today. You can read all about it &lt;a href=&quot;http://downloads.asterisk.org/pub/security/AST-2011-001.html&quot; rel=&quot;nofollow&quot;&gt;here&lt;/a&gt;.

We have developed a script which will quickly patch your system and eliminate the problem. Log into your server as root and issue the following commands:

cd /root
wget http://incrediblepbx.com/sipfix
chmod +x sipfix
./sipfix

Please apply this patch immediately to protect your server!]]></description>
			<content:encoded><![CDATA[<p>A major SIP security vulnerability was discovered in all versions of Asterisk today. You can read all about it <a href="http://downloads.asterisk.org/pub/security/AST-2011-001.html" rel="nofollow">here</a>.</p>
<p>We have developed a script which will quickly patch your system and eliminate the problem. Log into your server as root and issue the following commands:</p>
<p>cd /root<br />
wget <a href="http://incrediblepbx.com/sipfix" rel="nofollow ugc">http://incrediblepbx.com/sipfix</a><br />
chmod +x sipfix<br />
./sipfix</p>
<p>Please apply this patch immediately to protect your server!</p>
]]></content:encoded>
		
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		<title>
		By: Suraj Joneja		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13453</link>

		<dc:creator><![CDATA[Suraj Joneja]]></dc:creator>
		<pubDate>Mon, 17 Jan 2011 17:14:34 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13453</guid>

					<description><![CDATA[Thank you so very much!

Everything worked just as you described! It was plug and play! This is my first time with Linux, although I have been spending some time on pbxes.org.

Google voice is working fine too!

I have registered on the PBXinaflash forums and am still in the moderation queue and can&#039;t post my questions.

I have gone through your &quot;Remote Phone Meets the Travelin’ Man&quot; article and I have a requirement to allow dynamic IP users in my organization to connect to extensions, probably without the method you mention. Although you&#039;d say its rather insecure but I saw that your implmentation has fail2ban (although you replied that it is not realtime) I still wish to allow any IP to connect to the IPPBX without the authentication.

WIll post other queries when I am allowed on the forums!

Thanks a lot once again!]]></description>
			<content:encoded><![CDATA[<p>Thank you so very much!</p>
<p>Everything worked just as you described! It was plug and play! This is my first time with Linux, although I have been spending some time on pbxes.org.</p>
<p>Google voice is working fine too!</p>
<p>I have registered on the PBXinaflash forums and am still in the moderation queue and can&#8217;t post my questions.</p>
<p>I have gone through your "Remote Phone Meets the Travelin’ Man" article and I have a requirement to allow dynamic IP users in my organization to connect to extensions, probably without the method you mention. Although you&#8217;d say its rather insecure but I saw that your implmentation has fail2ban (although you replied that it is not realtime) I still wish to allow any IP to connect to the IPPBX without the authentication.</p>
<p>WIll post other queries when I am allowed on the forums!</p>
<p>Thanks a lot once again!</p>
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		<title>
		By: jbrukardt		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13436</link>

		<dc:creator><![CDATA[jbrukardt]]></dc:creator>
		<pubDate>Tue, 11 Jan 2011 15:41:01 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13436</guid>

					<description><![CDATA[Glad my post over in the forums spurred a bit of feature add for those of us who like google voice :D

Thank you for all the hard work on this Ward, your constant patience and excellent guides have been very informative and helpful both.]]></description>
			<content:encoded><![CDATA[<p>Glad my post over in the forums spurred a bit of feature add for those of us who like google voice 😀</p>
<p>Thank you for all the hard work on this Ward, your constant patience and excellent guides have been very informative and helpful both.</p>
]]></content:encoded>
		
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		<title>
		By: ward		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13419</link>

		<dc:creator><![CDATA[ward]]></dc:creator>
		<pubDate>Sun, 09 Jan 2011 12:47:17 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13419</guid>

					<description><![CDATA[For unanswered calls, if you&#039;d prefer to use Google Voice voicemail (with transcription) rather than Asterisk voicemail, here&#039;s a thread on the &lt;a href=&quot;http://nerd.bz/dSwanz&quot; rel=&quot;nofollow&quot;&gt;PIAF Forums&lt;/a&gt; that will show you how to make this simple change.]]></description>
			<content:encoded><![CDATA[<p>For unanswered calls, if you&#8217;d prefer to use Google Voice voicemail (with transcription) rather than Asterisk voicemail, here&#8217;s a thread on the <a href="http://nerd.bz/dSwanz" rel="nofollow">PIAF Forums</a> that will show you how to make this simple change.</p>
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		<title>
		By: Steve		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13370</link>

		<dc:creator><![CDATA[Steve]]></dc:creator>
		<pubDate>Wed, 29 Dec 2010 23:38:48 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13370</guid>

					<description><![CDATA[This is awesome! 
Is there an Aastra 6757i configuration file that will work with this? It&#039;s mostly working with my old config after I changed the IP of the server, but vm doesn&#039;t connect.]]></description>
			<content:encoded><![CDATA[<p>This is awesome!<br />
Is there an Aastra 6757i configuration file that will work with this? It&#8217;s mostly working with my old config after I changed the IP of the server, but vm doesn&#8217;t connect.</p>
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		<title>
		By: TK		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13351</link>

		<dc:creator><![CDATA[TK]]></dc:creator>
		<pubDate>Sun, 26 Dec 2010 19:13:08 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13351</guid>

					<description><![CDATA[Hello,

Thanks so much for taking the time to put together those tutorial. I am a total newbie and was able to follow through without a glitch.

Everything went extremely smooth until the sip phone step ... I have installed EKIGA on ubuntu 10.04 but i keep getting a &quot;could not register (failed)&quot; error.

I have tried several other softphone clients to no avail ...

Does anybody has any ideas or suggestions?]]></description>
			<content:encoded><![CDATA[<p>Hello,</p>
<p>Thanks so much for taking the time to put together those tutorial. I am a total newbie and was able to follow through without a glitch.</p>
<p>Everything went extremely smooth until the sip phone step &#8230; I have installed EKIGA on ubuntu 10.04 but i keep getting a "could not register (failed)" error.</p>
<p>I have tried several other softphone clients to no avail &#8230;</p>
<p>Does anybody has any ideas or suggestions?</p>
]]></content:encoded>
		
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		<title>
		By: BlaSTiWi		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13321</link>

		<dc:creator><![CDATA[BlaSTiWi]]></dc:creator>
		<pubDate>Mon, 20 Dec 2010 15:08:24 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13321</guid>

					<description><![CDATA[Got everything going again w/ GV using the latest ISO etc ... Tkx so much! But I&#039;m having problem w/ the VM, is said &#039;an error has occured&#039; and failed to save even after several diff. installs, various msg length, between ext. or from outside, rebuild ext. 701/702 VM setup or even on new ext. w/o pre-setup VM.

&lt;i&gt;[WM: Take advantage of the &lt;a href=&quot;http://pbxinaflash.com/forum/&quot; rel=&quot;nofollow&quot;&gt;PIAF Forums&lt;/a&gt;. It&#039;s not only free, but there are hundreds of gurus willing to help.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Got everything going again w/ GV using the latest ISO etc &#8230; Tkx so much! But I&#8217;m having problem w/ the VM, is said &#8216;an error has occured&#8217; and failed to save even after several diff. installs, various msg length, between ext. or from outside, rebuild ext. 701/702 VM setup or even on new ext. w/o pre-setup VM.</p>
<p><i>[WM: Take advantage of the <a href="http://pbxinaflash.com/forum/" rel="nofollow">PIAF Forums</a>. It&#8217;s not only free, but there are hundreds of gurus willing to help.]</i></p>
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		<title>
		By: PeterM		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13303</link>

		<dc:creator><![CDATA[PeterM]]></dc:creator>
		<pubDate>Thu, 16 Dec 2010 03:42:07 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13303</guid>

					<description><![CDATA[Sorry to hear about your hosting provider. I&#039;m trying to get a friend to move from Omnis as well, after similarly ridiculous treatment. We&#039;re looking at Web Faction, but that&#039;s neither here nor there. I&#039;m so psyched to install the new version. Great work, guys!

&lt;i&gt;[WM: Actually, omnis.com is only the registrar for our domains. But we&#039;re still dependent upon them not to break DNS. :roll: ]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Sorry to hear about your hosting provider. I&#8217;m trying to get a friend to move from Omnis as well, after similarly ridiculous treatment. We&#8217;re looking at Web Faction, but that&#8217;s neither here nor there. I&#8217;m so psyched to install the new version. Great work, guys!</p>
<p><i>[WM: Actually, omnis.com is only the registrar for our domains. But we&#8217;re still dependent upon them not to break DNS. 🙄 ]</i></p>
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		<title>
		By: TonyN		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13299</link>

		<dc:creator><![CDATA[TonyN]]></dc:creator>
		<pubDate>Wed, 15 Dec 2010 17:33:00 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13299</guid>

					<description><![CDATA[Was Gizmo Integration sorted out in this release?

&lt;i&gt;[WM: That&#039;s a problem at Google&#039;s end unfortunately. Sorry.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Was Gizmo Integration sorted out in this release?</p>
<p><i>[WM: That&#8217;s a problem at Google&#8217;s end unfortunately. Sorry.]</i></p>
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		<title>
		By: George		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13298</link>

		<dc:creator><![CDATA[George]]></dc:creator>
		<pubDate>Wed, 15 Dec 2010 16:45:05 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13298</guid>

					<description><![CDATA[First, thanks for all your work on this!

I keep on having a problem using PIAF with google voice. When I call using X-Lite or Zoiper through PIAF to a person on a cell or landline in the states the recipient of the call on the cell or landline hears himself twice or hears an echo whereas the person on softphone does not hear any echo. I am running PIAF on a powerful laptop over broadband. Any advice would be much appreciated.]]></description>
			<content:encoded><![CDATA[<p>First, thanks for all your work on this!</p>
<p>I keep on having a problem using PIAF with google voice. When I call using X-Lite or Zoiper through PIAF to a person on a cell or landline in the states the recipient of the call on the cell or landline hears himself twice or hears an echo whereas the person on softphone does not hear any echo. I am running PIAF on a powerful laptop over broadband. Any advice would be much appreciated.</p>
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		<title>
		By: kumar		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13294</link>

		<dc:creator><![CDATA[kumar]]></dc:creator>
		<pubDate>Tue, 14 Dec 2010 20:15:10 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13294</guid>

					<description><![CDATA[Is a 32bit openvz template available for this edition of Incredible PBX?

&lt;i&gt;[WM: Not yet, but it&#039;s on the Wish List.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Is a 32bit openvz template available for this edition of Incredible PBX?</p>
<p><i>[WM: Not yet, but it&#8217;s on the Wish List.]</i></p>
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		<title>
		By: Molotof		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13292</link>

		<dc:creator><![CDATA[Molotof]]></dc:creator>
		<pubDate>Tue, 14 Dec 2010 17:24:31 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13292</guid>

					<description><![CDATA[I don&#039;t know how to thank you for your on going effort in this push button magic! You are really wonderful in all that you do, bringing me joy for years, every time i visit your page i find my solution already there ripe and ready for the picking ! 

Thank you so much for a nice and flawless Google voice fix and as always a beautiful pbx in a flash script !]]></description>
			<content:encoded><![CDATA[<p>I don&#8217;t know how to thank you for your on going effort in this push button magic! You are really wonderful in all that you do, bringing me joy for years, every time i visit your page i find my solution already there ripe and ready for the picking ! </p>
<p>Thank you so much for a nice and flawless Google voice fix and as always a beautiful pbx in a flash script !</p>
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		<title>
		By: Dishjuarez		</title>
		<link>https://nerdvittles.com/incredible-pbx-for-asterisk-181-back-from-the-brink/comment-page-1/#comment-13291</link>

		<dc:creator><![CDATA[Dishjuarez]]></dc:creator>
		<pubDate>Tue, 14 Dec 2010 04:44:34 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=712#comment-13291</guid>

					<description><![CDATA[Saludos, todo esta trabajando bien con mi numero de El Paso Tx, Regards all is ok, working very well with my number from El Paso Tx, thanks to all team of PIAF, this is a little marvel and i dont know nothing to compare with this, Congratulations!!!]]></description>
			<content:encoded><![CDATA[<p>Saludos, todo esta trabajando bien con mi numero de El Paso Tx, Regards all is ok, working very well with my number from El Paso Tx, thanks to all team of PIAF, this is a little marvel and i dont know nothing to compare with this, Congratulations!!!</p>
]]></content:encoded>
		
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