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	<title>Comments on: Jumping Onto the VoIP Bandwagon</title>
	<atom:link href="http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/feed/" rel="self" type="application/rss+xml" />
	<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/</link>
	<description>Christianity, Orthodoxy and Technology</description>
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		<title>By: Using SpinVox with Asterisk &#187; anchorite.org</title>
		<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/comment-page-1/#comment-9756</link>
		<dc:creator>Using SpinVox with Asterisk &#187; anchorite.org</dc:creator>
		<pubDate>Wed, 28 Mar 2007 18:02:39 +0000</pubDate>
		<guid isPermaLink="false">http://www.anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/#comment-9756</guid>
		<description>[...] Per Matt Singley&#8217;s recommendation, I decided to give SpinVox a try. I don&#8217;t use voicemail on my cell phone so I decided to run it on my office line. Setting it up with our Asterisk PBX was pretty straightforward. [...]</description>
		<content:encoded><![CDATA[<p>[...] Per Matt Singley&#8217;s recommendation, I decided to give SpinVox a try. I don&#8217;t use voicemail on my cell phone so I decided to run it on my office line. Setting it up with our Asterisk PBX was pretty straightforward. [...]</p>
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		<title>By: VoIP PBX Appliances &#187; anchorite.org</title>
		<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/comment-page-1/#comment-7292</link>
		<dc:creator>VoIP PBX Appliances &#187; anchorite.org</dc:creator>
		<pubDate>Mon, 19 Mar 2007 17:40:25 +0000</pubDate>
		<guid isPermaLink="false">http://www.anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/#comment-7292</guid>
		<description>[...] For those of you who are interested in an open source VoIP PBX but don&#8217;t have the time or desire to build your own Asterisk box, Fonality is releasing the Trixbox Appliance. Trixbox is the new name for Asterisk@Home, the platform we currently run at church. Asterisk@Home 2.7 is working well for us, although we do look forward to upgrading soon so that we can take advantage of the new features in trixbox. [...]</description>
		<content:encoded><![CDATA[<p>[...] For those of you who are interested in an open source VoIP PBX but don&#8217;t have the time or desire to build your own Asterisk box, Fonality is releasing the Trixbox Appliance. Trixbox is the new name for Asterisk@Home, the platform we currently run at church. Asterisk@Home 2.7 is working well for us, although we do look forward to upgrading soon so that we can take advantage of the new features in trixbox. [...]</p>
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		<title>By: Andrew Mitry</title>
		<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/comment-page-1/#comment-25</link>
		<dc:creator>Andrew Mitry</dc:creator>
		<pubDate>Thu, 13 Jul 2006 04:32:08 +0000</pubDate>
		<guid isPermaLink="false">http://www.anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/#comment-25</guid>
		<description>You may want to still get a closer look at Asterisk, it can integrate with your existing PBX and provide a lot of additonal functionality at a fraction of the cost (I know those traditional systems charge a pretty penny for every feature upgrade or addon).  Check out the &lt;a rel=&quot;nofollow&quot; title=&quot;Trixbox Virtual Appliance&quot; href=&quot;http://www.vmware.com/vmtn/appliances/directory/49&quot; rel=&quot;nofollow&quot;&gt; Trixbox Virtual Appliance&lt;/a&gt; for VMware Server.</description>
		<content:encoded><![CDATA[<p>You may want to still get a closer look at Asterisk, it can integrate with your existing PBX and provide a lot of additonal functionality at a fraction of the cost (I know those traditional systems charge a pretty penny for every feature upgrade or addon).  Check out the <a rel="nofollow" title="Trixbox Virtual Appliance" href="http://www.vmware.com/vmtn/appliances/directory/49" rel="nofollow"> Trixbox Virtual Appliance</a> for VMware Server.</p>
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		<title>By: Jason Powell</title>
		<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/comment-page-1/#comment-24</link>
		<dc:creator>Jason Powell</dc:creator>
		<pubDate>Thu, 13 Jul 2006 03:51:58 +0000</pubDate>
		<guid isPermaLink="false">http://www.anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/#comment-24</guid>
		<description>Great post!!!

If we had understood phone stuff better we would have looked more closely at Asterisk last fall.</description>
		<content:encoded><![CDATA[<p>Great post!!!</p>
<p>If we had understood phone stuff better we would have looked more closely at Asterisk last fall.</p>
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		<title>By: Rolf</title>
		<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/comment-page-1/#comment-22</link>
		<dc:creator>Rolf</dc:creator>
		<pubDate>Wed, 12 Jul 2006 01:44:46 +0000</pubDate>
		<guid isPermaLink="false">http://www.anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/#comment-22</guid>
		<description>I&#039;m using the &lt;a href=&quot;http://www.digium.com/en/products/hardware/te110p.php&quot; rel=&quot;nofollow&quot;&gt;Digium 110P&lt;/a&gt;. We have it in a HP DL360 with 4GB of  RAM, dual 2.8G procs, and two 80G SATA drives.  The size of the server is overkill, but we&#039;re not sure how quickly we&#039;ll grow, so we wanted to be safe :).</description>
		<content:encoded><![CDATA[<p>I&#8217;m using the <a href="http://www.digium.com/en/products/hardware/te110p.php" rel="nofollow">Digium 110P</a>. We have it in a HP DL360 with 4GB of  RAM, dual 2.8G procs, and two 80G SATA drives.  The size of the server is overkill, but we&#8217;re not sure how quickly we&#8217;ll grow, so we wanted to be safe <img src='http://anchorite.org/blog/wp-includes/images/smilies/icon_smile.gif' alt=':)' class='wp-smiley' /> .</p>
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		<title>By: Andrew Mitry</title>
		<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/comment-page-1/#comment-21</link>
		<dc:creator>Andrew Mitry</dc:creator>
		<pubDate>Tue, 11 Jul 2006 21:09:53 +0000</pubDate>
		<guid isPermaLink="false">http://www.anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/#comment-21</guid>
		<description>Jason, you are welcome for the answers, keep us posted on how the move to VoIP goes.  Let me know if you have any other questions.

Rolf, which T1 card are you using, the Digium or the Sangoma?</description>
		<content:encoded><![CDATA[<p>Jason, you are welcome for the answers, keep us posted on how the move to VoIP goes.  Let me know if you have any other questions.</p>
<p>Rolf, which T1 card are you using, the Digium or the Sangoma?</p>
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		<title>By: Rolf Brusletto</title>
		<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/comment-page-1/#comment-20</link>
		<dc:creator>Rolf Brusletto</dc:creator>
		<pubDate>Tue, 11 Jul 2006 19:37:13 +0000</pubDate>
		<guid isPermaLink="false">http://www.anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/#comment-20</guid>
		<description>Very good article! I too come from a past, installing and maintaining digital key systems(Toshiba mostly), and Asterisk @ Home(pre trixbox) was a great opportunity for us to take advantage of. We use Polycom IP501&#039;s and some IP 4000&#039;s. Couple with the meetme features, and other great features, we&#039;ve been nothing less than ecstatic. One thing I can definitely attest to is having the voip network on it&#039;s own vlan or network segment. We chose that route after noticing some echo on our side, but not at the far clients side. We use a single t1 card w/ a full t1, and 80 dids from Qwest without a glitch.</description>
		<content:encoded><![CDATA[<p>Very good article! I too come from a past, installing and maintaining digital key systems(Toshiba mostly), and Asterisk @ Home(pre trixbox) was a great opportunity for us to take advantage of. We use Polycom IP501&#8242;s and some IP 4000&#8242;s. Couple with the meetme features, and other great features, we&#8217;ve been nothing less than ecstatic. One thing I can definitely attest to is having the voip network on it&#8217;s own vlan or network segment. We chose that route after noticing some echo on our side, but not at the far clients side. We use a single t1 card w/ a full t1, and 80 dids from Qwest without a glitch.</p>
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		<title>By: Jason</title>
		<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/comment-page-1/#comment-19</link>
		<dc:creator>Jason</dc:creator>
		<pubDate>Tue, 11 Jul 2006 16:25:38 +0000</pubDate>
		<guid isPermaLink="false">http://www.anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/#comment-19</guid>
		<description>Thanks for the answers!

I absolutely love my E61.  Voice quality can be choppy from time to time, but that could be my lousy AP, but that aside, it works great.  From time to time, it&#039;s nice to hit the cfwd button on my desk phone and roam around the house..  I also get my email on it via BlackBerry Connect, so it&#039;s good...</description>
		<content:encoded><![CDATA[<p>Thanks for the answers!</p>
<p>I absolutely love my E61.  Voice quality can be choppy from time to time, but that could be my lousy AP, but that aside, it works great.  From time to time, it&#8217;s nice to hit the cfwd button on my desk phone and roam around the house..  I also get my email on it via BlackBerry Connect, so it&#8217;s good&#8230;</p>
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		<title>By: Andrew Mitry</title>
		<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/comment-page-1/#comment-18</link>
		<dc:creator>Andrew Mitry</dc:creator>
		<pubDate>Tue, 11 Jul 2006 00:31:26 +0000</pubDate>
		<guid isPermaLink="false">http://www.anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/#comment-18</guid>
		<description>We actually purchased the Linksys 941&#039;s just as they began shipping, the 942&#039;s were not even announced at the time (there were some rumors floating around, but nothing confirmed).  As we add I do plan on purchasing the 922&#039;s or 942&#039;s, I am especially interested in the integrated switch as that it supposedly supports VLan tagging.  I stuck with the 2-lines, no need for the 4-line upgrade in our configuration, each user can enable call waiting on their extension if they so choose and can dial out a second call on the same extension.  The only time I configured the second extension is if two people were sharing the same phone, I would give them seperate extensions.  As for POE, none of our current switches support it and POE injectors with multiple ports are pretty expensive.  I am considering purchasing a POE switch that runs -48 DC to put on our telecom rack (it has an 8-hour backup, Cox installed it when they put in the fiber that our Data and PRI come across).  I would then strategically place the newer phones I buy in areas that would be important to have a working phone during a power outage.

How is the Nokia E61?  I have been eyeing it for a while, but we are tied to Sprint for at least another year.  I have tested the Linksys WIP300 and it works great, we are going to roll it out as a&quot;cordless&quot; phone that covers the whole facility once we finish installing all the wireless access points.

You are correct about the DIDs coming down the channels on the PRI, there are 23 channels for voice and one for data (transmits caller ID and call setup info).  Usually if you are recieiving incoming calls you want to get at least one block of 20 DIDs that will include your main number.   A block of 20 DIDs typically costs between $10 and $20 a month extra (I think we are paying $10, but I need to double-check).

I hope I answered most of your questions.  God bless you too!</description>
		<content:encoded><![CDATA[<p>We actually purchased the Linksys 941&#8242;s just as they began shipping, the 942&#8242;s were not even announced at the time (there were some rumors floating around, but nothing confirmed).  As we add I do plan on purchasing the 922&#8242;s or 942&#8242;s, I am especially interested in the integrated switch as that it supposedly supports VLan tagging.  I stuck with the 2-lines, no need for the 4-line upgrade in our configuration, each user can enable call waiting on their extension if they so choose and can dial out a second call on the same extension.  The only time I configured the second extension is if two people were sharing the same phone, I would give them seperate extensions.  As for POE, none of our current switches support it and POE injectors with multiple ports are pretty expensive.  I am considering purchasing a POE switch that runs -48 DC to put on our telecom rack (it has an 8-hour backup, Cox installed it when they put in the fiber that our Data and PRI come across).  I would then strategically place the newer phones I buy in areas that would be important to have a working phone during a power outage.</p>
<p>How is the Nokia E61?  I have been eyeing it for a while, but we are tied to Sprint for at least another year.  I have tested the Linksys WIP300 and it works great, we are going to roll it out as a&#8221;cordless&#8221; phone that covers the whole facility once we finish installing all the wireless access points.</p>
<p>You are correct about the DIDs coming down the channels on the PRI, there are 23 channels for voice and one for data (transmits caller ID and call setup info).  Usually if you are recieiving incoming calls you want to get at least one block of 20 DIDs that will include your main number.   A block of 20 DIDs typically costs between $10 and $20 a month extra (I think we are paying $10, but I need to double-check).</p>
<p>I hope I answered most of your questions.  God bless you too!</p>
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		<title>By: Jason</title>
		<link>http://anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/comment-page-1/#comment-17</link>
		<dc:creator>Jason</dc:creator>
		<pubDate>Tue, 11 Jul 2006 00:05:56 +0000</pubDate>
		<guid isPermaLink="false">http://www.anchorite.org/blog/2006/07/07/jumping-onto-the-voip-bandwagon/#comment-17</guid>
		<description>What, if I may ask, caused you to go with the Linksys 941&#039;s over the 942&#039;s?  Was it just the $1k you saved on the 20 stations?  Did you just stick with the 2 lines, or did you go for the 4-line upgrade?  Not interested in PoE, I guess, as well..

Our church is considering a move to VoIP, and I&#039;m the geek in charge of the investigation.  Thus far, I&#039;ve installed in my home, a Trixbox (running in a vm under vmware server on my home server), connected to a handful of SIP/IAX trunks (Gizmo, FWD, Voipdiscount, Voipjet, and a SPA-3000 for PSTN connectivity).  My desk phone is a Linksys SPA-942, I&#039;ve got a cordless phone connected to the FXS port of the SPA-3000 and I&#039;ve got my Nokia E61 talking to the PBX over WiFi.  Go ahead, call me a geek. :)

If you&#039;re doing multiple lines for the users, how are you handling extension assignments (200/1200 or 200/201, etc?).

Further, as I am not skilled in the ways of the PRI, am I correct in thinking that you&#039;ve got 23 channels per PRI, which are not directly mapped to any particular DID?  In other words, a call to DID1 may come in on channel0, but the next call to DID1 may come in on channel7?  If so, does it typically cost extra to get additional DIDs assigned to  a PRI, i.e. you get a single # associated with the PRI, then add extra DID #&#039;s as an add-on?  Am I making sense?

Ok, that&#039;s a lot of questions to throw your way..  Thanks and God bless!</description>
		<content:encoded><![CDATA[<p>What, if I may ask, caused you to go with the Linksys 941&#8242;s over the 942&#8242;s?  Was it just the $1k you saved on the 20 stations?  Did you just stick with the 2 lines, or did you go for the 4-line upgrade?  Not interested in PoE, I guess, as well..</p>
<p>Our church is considering a move to VoIP, and I&#8217;m the geek in charge of the investigation.  Thus far, I&#8217;ve installed in my home, a Trixbox (running in a vm under vmware server on my home server), connected to a handful of SIP/IAX trunks (Gizmo, FWD, Voipdiscount, Voipjet, and a SPA-3000 for PSTN connectivity).  My desk phone is a Linksys SPA-942, I&#8217;ve got a cordless phone connected to the FXS port of the SPA-3000 and I&#8217;ve got my Nokia E61 talking to the PBX over WiFi.  Go ahead, call me a geek. <img src='http://anchorite.org/blog/wp-includes/images/smilies/icon_smile.gif' alt=':)' class='wp-smiley' /> </p>
<p>If you&#8217;re doing multiple lines for the users, how are you handling extension assignments (200/1200 or 200/201, etc?).</p>
<p>Further, as I am not skilled in the ways of the PRI, am I correct in thinking that you&#8217;ve got 23 channels per PRI, which are not directly mapped to any particular DID?  In other words, a call to DID1 may come in on channel0, but the next call to DID1 may come in on channel7?  If so, does it typically cost extra to get additional DIDs assigned to  a PRI, i.e. you get a single # associated with the PRI, then add extra DID #&#8217;s as an add-on?  Am I making sense?</p>
<p>Ok, that&#8217;s a lot of questions to throw your way..  Thanks and God bless!</p>
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