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		<title>Voxilla VoIP Forum</title>
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			<title>503 error in SPA3102</title>
			<link>http://forum.voxilla.com/linksys-sipura-voip-support-forum/503-error-spa3102-40353.html</link>
			<pubDate>Sun, 22 Nov 2009 15:00:30 GMT</pubDate>
			<description>Thanks Hwittenb  for your post.  
    I have upgraded the SPA3102 to the latest version as you have suggested. 
 
Unfortunately i would not be able...</description>
			<content:encoded><![CDATA[<div>Thanks Hwittenb  for your post. <br />
    I have upgraded the SPA3102 to the latest version as you have suggested.<br />
<br />
Unfortunately i would not be able to immediately post the results. <br />
<br />
I will also try and disable the Line1 connection or the fallback before trying the call out once again. <br />
I jut hope it would work this time around. <br />
Will post the results after 2 days and keep you updated.</div>


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</div>]]></content:encoded>
			<category domain="http://forum.voxilla.com/linksys-sipura-voip-support-forum/">Linksys (Sipura) VoIP Support Forum</category>
			<dc:creator>darkhorse</dc:creator>
			<guid isPermaLink="true">http://forum.voxilla.com/linksys-sipura-voip-support-forum/503-error-spa3102-40353.html</guid>
		</item>
		<item>
			<title>video streaming is nt allow</title>
			<link>http://forum.voxilla.com/skype-support-forum/video-streaming-nt-allow-40364.html</link>
			<pubDate>Sun, 22 Nov 2009 12:12:28 GMT</pubDate>
			<description>i have been using skype version 4.1.0.141,,  recently i have change my version to 4.1.0.179.. then i could not use the web cam ,, as the screen...</description>
			<content:encoded><![CDATA[<div>i have been using skype version 4.1.0.141,,  recently i have change my version to 4.1.0.179.. then i could not use the web cam ,, as the screen prompt (  streaming is not allow in this mode..)  can help?:confused:</div>


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			<category domain="http://forum.voxilla.com/skype-support-forum/">Skype Support Forum</category>
			<dc:creator>dennis_q11</dc:creator>
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		<item>
			<title>SPA3102 and Vonage setup</title>
			<link>http://forum.voxilla.com/linksys-sipura-voip-support-forum/spa3102-vonage-setup-37634.html</link>
			<pubDate>Sun, 22 Nov 2009 07:54:46 GMT</pubDate>
			<description>First i apologize for the late reply... And then the good news.. everything is working wonderfully, 
 
pstn2voip 
voip2pstn 
 
Thanks a lot to...</description>
			<content:encoded><![CDATA[<div>First i apologize for the late reply... And then the good news.. everything is working wonderfully,<br />
<br />
pstn2voip<br />
voip2pstn<br />
<br />
Thanks a lot to hwitten.. Now everytime i talk to my home... a bug thanks would be to you...<br />
<br />
Anand</div>


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			<category domain="http://forum.voxilla.com/linksys-sipura-voip-support-forum/">Linksys (Sipura) VoIP Support Forum</category>
			<dc:creator>anthiyag</dc:creator>
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		</item>
		<item>
			<title>Daily Updates of BrainTel (Service Status)</title>
			<link>http://forum.voxilla.com/braintel-support-forum/daily-updates-of-braintel-service-status-35891.html</link>
			<pubDate>Sun, 22 Nov 2009 05:57:29 GMT</pubDate>
			<description>Slam Guys 
now a days i am getting problem with my BT line, Voice Breakdown while talking through CAD System, 
have any one similar then please share...</description>
			<content:encoded><![CDATA[<div>Slam Guys<br />
now a days i am getting problem with my BT line, Voice Breakdown while talking through CAD System,<br />
have any one similar then please share otherwise it seems to be problem at my end only</div>


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</div>]]></content:encoded>
			<category domain="http://forum.voxilla.com/braintel-support-forum/">BrainTEL Support Forum</category>
			<dc:creator>asifhafeez</dc:creator>
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		</item>
		<item>
			<title>SPA3102</title>
			<link>http://forum.voxilla.com/linksys-sipura-voip-support-forum/spa3102-40352.html</link>
			<pubDate>Sun, 22 Nov 2009 01:09:13 GMT</pubDate>
			<description><![CDATA[Helo Guys, 
I'm zero knoweldge on networking an routers and all that stuff, I learned about the SPA3102 as disperately searching for a way to cut...]]></description>
			<content:encoded><![CDATA[<div>Helo Guys,<br />
I'm zero knoweldge on networking an routers and all that stuff, I learned about the SPA3102 as disperately searching for a way to cut down my monthly international bill (350  $+),<br />
I now have :<br />
 <br />
*Two SPA3102<br />
*Free callcentric accounts, Currently i have three of them, I can add more<br />
I'm setting up one to station here in USA, with AT&amp;T Uverse service and a PSTN Line, I temporarily setup the other SPA3102 at the same location where the firsst one is for purpose of figuring it out.<br />
 <br />
I achieved:<br />
* Call in the PSTN line , gives me a VOIP Dialtone &quot; Problem 1, 2&quot;<br />
*Call the voip line on the SPA3102 gives me a PSTN dialtone, I can make calls successfully &quot; Problem 3&quot;<br />
 <br />
Problem 1: When i call the VOIP account , that is asssigned to the temporary SPA, This callcentric message come out&quot; the person you are trying to reach is currently unavailable&quot;<br />
 <br />
Problem 2: I cahnged the port on the temporary SPA SIP tab to 4800 and it is working, only once, and then the message come again, I guess the voip is not hanging up?<br />
 <br />
Problem 3: When the pstn call is established, after forwarding it from the voip, the other side on the pstn line&quot; my wife&quot;, hears a VOIP dial tone and then hangs up&quot;<br />
 <br />
Problem 4 : The pstn call oesn't hang up after i hang up the VOIP from the temporary SPA</div>


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			<category domain="http://forum.voxilla.com/linksys-sipura-voip-support-forum/">Linksys (Sipura) VoIP Support Forum</category>
			<dc:creator>ahussein</dc:creator>
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		</item>
		<item>
			<title><![CDATA[How to get working inbound and outbound via SPA9000&SPA400 ?]]></title>
			<link>http://forum.voxilla.com/linksys-spa9000-spa400-support-forum/get-working-inbound-outbound-via-spa9000-spa400-33317.html</link>
			<pubDate>Sun, 22 Nov 2009 00:13:58 GMT</pubDate>
			<description><![CDATA[I'm trying to get this to work too!  I have 4 incoming lines plugged into my SPA400. 
 
PSTN-1 -> Regular Business Line 
PSTN-2 -> Regular Business...]]></description>
			<content:encoded><![CDATA[<div>I'm trying to get this to work too!  I have 4 incoming lines plugged into my SPA400.<br />
<br />
PSTN-1 -&gt; Regular Business Line<br />
PSTN-2 -&gt; Regular Business Line<br />
PSTN-3 -&gt; Regular Business Line<br />
PSTN-4 -&gt; Fax Line<br />
<br />
The SPA9000 is configured to the SPA400.  When a call comes in, they are handled by this:<br />
<br />
Contact List: aa<br />
<br />
The problem is, Fax calls coming in are being greeted by the Auto-Attendant.  They should be pumped through to the FXS line on the SPA9000 so that they will go directly to the fax machine.  The reason I've plugged the Fax line into the SPA400 is because we are very busy sometimes, and occasionally need a line 4 to call out.  So it's convenient having it, but I cannot have fax calls coming into the AA.  According to what I've seen above, I should change it to:<br />
<br />
Contact List: FAX:400 | aa<br />
<br />
On the SPA400 change the name of the gateway port for line 3 to FAX.  I've tried that, but it does not seem to work.  Does anyone know what I'm doing wrong?  Or does anyone else have this set up and working?  Your help would be immensely appreciated.<br />
<br />
Thanks. :D</div>


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			<category domain="http://forum.voxilla.com/linksys-spa9000-spa400-support-forum/">Linksys SPA9000/SPA400 Support Forum</category>
			<dc:creator>Kevin270</dc:creator>
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		<item>
			<title>Tricks for getting PAP2T to talk to SPA3102 over internet?</title>
			<link>http://forum.voxilla.com/linksys-sipura-voip-support-forum/tricks-getting-pap2t-talk-spa3102-over-internet-35286.html</link>
			<pubDate>Sat, 21 Nov 2009 21:45:09 GMT</pubDate>
			<description>I really appreciate your help, hwittenb. 
 
I went to menu voip-to-pstn on the SPA3102 and changed the default dial plan from (xx.) to the one you...</description>
			<content:encoded><![CDATA[<div>I really appreciate your help, hwittenb.<br />
<br />
I went to menu voip-to-pstn on the SPA3102 and changed the default dial plan from (xx.) to the one you said ([x*][x*].) and it is now accepting my dialing with * characters. <br />
<br />
Thank you very much. <br />
<br />
Best regards,<br />
Flavio</div>


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			<category domain="http://forum.voxilla.com/linksys-sipura-voip-support-forum/">Linksys (Sipura) VoIP Support Forum</category>
			<dc:creator>fca1973</dc:creator>
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		</item>
		<item>
			<title>How I can forward PSTN to VOIP and vice versa linksys 3102</title>
			<link>http://forum.voxilla.com/linksys-sipura-voip-support-forum/can-forward-pstn-voip-vice-versa-linksys-3102-a-32128.html</link>
			<pubDate>Sat, 21 Nov 2009 15:22:59 GMT</pubDate>
			<description>---Quote (Originally by mohnashaat)--- 
is there any advantage of changing it? cuz i saw on another thread that you changed the polarity 
---End...</description>
			<content:encoded><![CDATA[<div><div style="margin:20px; margin-top:5px; ">
	<div class="smallfont" style="margin-bottom:2px">Quote:</div>
	<table cellpadding="6" cellspacing="0" border="0" width="100%">
	<tr>
		<td class="alt2">
			<hr />
			
				<div>
					Originally Posted by <strong>mohnashaat</strong>
					(Post 135981)
				</div>
				<div style="font-style:italic">is there any advantage of changing it? cuz i saw on another thread that you changed the polarity</div>
			
			<hr />
		</td>
	</tr>
	</table>
</div>In that case I was getting a false CPC disconnect when the FXO port went off hook.  If you are getting a false CPC disconnect when the FXO port goes off hook there is another way to deal with it besides changing polarity.</div>


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</div>]]></content:encoded>
			<category domain="http://forum.voxilla.com/linksys-sipura-voip-support-forum/">Linksys (Sipura) VoIP Support Forum</category>
			<dc:creator>hwittenb</dc:creator>
			<guid isPermaLink="true">http://forum.voxilla.com/linksys-sipura-voip-support-forum/can-forward-pstn-voip-vice-versa-linksys-3102-a-32128.html</guid>
		</item>
		<item>
			<title>Can admin look into Certificate wizard?</title>
			<link>http://forum.voxilla.com/linksys-sipura-voip-support-forum/can-admin-look-into-certificate-wizard-39206.html</link>
			<pubDate>Sat, 21 Nov 2009 13:27:48 GMT</pubDate>
			<description>*****Bump*****</description>
			<content:encoded><![CDATA[<div>*****Bump*****</div>


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</div>]]></content:encoded>
			<category domain="http://forum.voxilla.com/linksys-sipura-voip-support-forum/">Linksys (Sipura) VoIP Support Forum</category>
			<dc:creator>rajil.s</dc:creator>
			<guid isPermaLink="true">http://forum.voxilla.com/linksys-sipura-voip-support-forum/can-admin-look-into-certificate-wizard-39206.html</guid>
		</item>
		<item>
			<title>HELP WITH HOW SKYPE WORKS PLEASE</title>
			<link>http://forum.voxilla.com/skype-support-forum/help-skype-works-please-40334.html</link>
			<pubDate>Sat, 21 Nov 2009 09:43:33 GMT</pubDate>
			<description>Hello, 
 
I live in the UK and downloaded Skype yesterday.  My Dad lives abroad and he downloaded Skype too. 
 
The question I have, is we have both...</description>
			<content:encoded><![CDATA[<div>Hello,<br />
<br />
I live in the UK and downloaded Skype yesterday.  My Dad lives abroad and he downloaded Skype too.<br />
<br />
The question I have, is we have both registered with Skype with our home phone numbers and would be talking to each other sat in front of our computers.<br />
<br />
If either one of us tried to call the other through Skype on the computer, would our home phones ring or how would we know to sit at the computer to talk to each other?<br />
<br />
Hope someone can help.<br />
<br />
Thanks in advance :)</div>


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</div>]]></content:encoded>
			<category domain="http://forum.voxilla.com/skype-support-forum/">Skype Support Forum</category>
			<dc:creator>Tinkerbelle</dc:creator>
			<guid isPermaLink="true">http://forum.voxilla.com/skype-support-forum/help-skype-works-please-40334.html</guid>
		</item>
		<item>
			<title>Pakistan India Bangladesh Nepa Egypt On Sale Good Oppertunity</title>
			<link>http://forum.voxilla.com/general-voip-discussion/pakistan-india-bangladesh-nepa-egypt-sale-good-oppertunity-40328.html</link>
			<pubDate>Sat, 21 Nov 2009 09:26:40 GMT</pubDate>
			<description>Hello, 
 
We Offer high and good Quality Routes to pak india bd nepal and egypt. 
 
We provide Fabulous offer to this routes for short period of...</description>
			<content:encoded><![CDATA[<div>Hello,<br />
<br />
We Offer high and good Quality Routes to pak india bd nepal and egypt.<br />
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<br />
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</div>]]></content:encoded>
			<category domain="http://forum.voxilla.com/general-voip-discussion/">General VoIP Discussion</category>
			<dc:creator>fouzia</dc:creator>
			<guid isPermaLink="true">http://forum.voxilla.com/general-voip-discussion/pakistan-india-bangladesh-nepa-egypt-sale-good-oppertunity-40328.html</guid>
		</item>
		<item>
			<title>Is IPKALL working</title>
			<link>http://forum.voxilla.com/ipkall-support-forum/ipkall-working-40308.html</link>
			<pubDate>Fri, 20 Nov 2009 20:11:42 GMT</pubDate>
			<description>What is the point of Ipkall still handing out numbers when the services is clearly not working. I have been trying for sometime since getting my...</description>
			<content:encoded><![CDATA[<div>What is the point of Ipkall still handing out numbers when the services is clearly not working. I have been trying for sometime since getting my number last week to ring my ipkall, but its dead! Also does anyone actually moderate this forum, as I sent a PM to IKPALL and got no response.. not good start really!</div>


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</div>]]></content:encoded>
			<category domain="http://forum.voxilla.com/ipkall-support-forum/">IPKall Support Forum</category>
			<dc:creator>wvista</dc:creator>
			<guid isPermaLink="true">http://forum.voxilla.com/ipkall-support-forum/ipkall-working-40308.html</guid>
		</item>
		<item>
			<title><![CDATA[Start your own VOIP+callingcard+callback+wholesale&retail business from your Home]]></title>
			<link>http://forum.voxilla.com/general-voip-discussion/start-your-own-voip-callingcard-callback-wholesale-retail-business-your-home-40303.html</link>
			<pubDate>Fri, 20 Nov 2009 18:05:55 GMT</pubDate>
			<description>Dear Friends:  
Setup ITSP company, a business under your own brand name? We have complete solution to launche ITSP (Internet Telephony Service...</description>
			<content:encoded><![CDATA[<div>Dear Friends: <br />
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</div>]]></content:encoded>
			<category domain="http://forum.voxilla.com/general-voip-discussion/">General VoIP Discussion</category>
			<dc:creator>LeadV0IP.com</dc:creator>
			<guid isPermaLink="true">http://forum.voxilla.com/general-voip-discussion/start-your-own-voip-callingcard-callback-wholesale-retail-business-your-home-40303.html</guid>
		</item>
		<item>
			<title>Call Center Software VoIP</title>
			<link>http://forum.voxilla.com/general-voip-discussion/call-center-software-voip-40300.html</link>
			<pubDate>Fri, 20 Nov 2009 17:52:24 GMT</pubDate>
			<description>Contact Center Online offers call center software for VoIP. Our company offers your first month rent for free, and we also offer leads when you rent...</description>
			<content:encoded><![CDATA[<div>Contact Center Online offers call center software for VoIP. Our company offers your first month rent for free, and we also offer leads when you rent a system from us!!! If you have questions, please email us at <a href="mailto:contactcenteronline@gmail.com">contactcenteronline@gmail.com</a> or call us at 214-257-8416.</div>


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</div>]]></content:encoded>
			<category domain="http://forum.voxilla.com/general-voip-discussion/">General VoIP Discussion</category>
			<dc:creator>contactcenteronline</dc:creator>
			<guid isPermaLink="true">http://forum.voxilla.com/general-voip-discussion/call-center-software-voip-40300.html</guid>
		</item>
		<item>
			<title>SPA3102 problem - needs rebooted once a day</title>
			<link>http://forum.voxilla.com/linksys-sipura-voip-support-forum/spa3102-problem-needs-rebooted-once-day-26878.html</link>
			<pubDate>Fri, 20 Nov 2009 15:56:51 GMT</pubDate>
			<description><![CDATA[I've just had this problem, and fixed it. I'm 99% sure most of these problems are the same.  
 
For me, and my parents using 2 different models of...]]></description>
			<content:encoded><![CDATA[<div>I've just had this problem, and fixed it. I'm 99% sure most of these problems are the same. <br />
<br />
For me, and my parents using 2 different models of router, it was fixed easily by setting up port triggering on your router set-up.<br />
<br />
For some reason, more modern routers have this problem, which I believe is something to do with the firewall. For me it was also not allowing any softphones to make voip calls too, but would allow the spa3102 or pap2t to register a first time, but then it would need a complete reboot of the router and linksys box to get it to re-register.<br />
<br />
To fix it, go to your router's set-up. Go to the port triggering option and add, for example:<br />
<br />
starting range 5060 to 5060<br />
finishing range 5060 to 5060<br />
enable - yes<br />
<br />
protocol: TCP/UDP<br />
<br />
If it's like mine, your spa/pap will now register by itself forever more!<br />
<br />
It has to be port triggering, port forwarding will not work.</div>


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