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	<title>
	Comments on: Follow-Me Roaming with Asterisk: Transparently Integrating Mobile Phones Into Your Dialplan	</title>
	<atom:link href="https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/feed/" rel="self" type="application/rss+xml" />
	<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/</link>
	<description>Ward Mundy&#039;s Technobabblelog</description>
	<lastBuildDate>Thu, 02 Jun 2011 14:11:02 +0000</lastBuildDate>
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		<title>
		By: LLL		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-9034</link>

		<dc:creator><![CDATA[LLL]]></dc:creator>
		<pubDate>Fri, 06 Mar 2009 20:56:52 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-9034</guid>

					<description><![CDATA[Voxee.com is also not accepting new accounts, but of course they don&#039;t mention that until you give them all of your contact info on their signup screen. Sounds like phishing to me. Anybody know what&#039;s up with them?]]></description>
			<content:encoded><![CDATA[<p>Voxee.com is also not accepting new accounts, but of course they don&#8217;t mention that until you give them all of your contact info on their signup screen. Sounds like phishing to me. Anybody know what&#8217;s up with them?</p>
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		<title>
		By: Bryan		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-5737</link>

		<dc:creator><![CDATA[Bryan]]></dc:creator>
		<pubDate>Fri, 09 Jan 2009 00:38:49 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-5737</guid>

					<description><![CDATA[Thank you for the info on setting up forward. I have done everything and it works fine. However the problem i have now is there is a 4-5 sec delay in the audio conversation. This is very difficult to talk with. When i place a call direct from one of my extensions there is almost zero delay. So it seems like it has to do with the forward. Any ideas? 

&lt;i&gt;[WM: Have you implemented the tips on Getting Rid of One-Way Audio in our &lt;a href=&quot;http://knol.pbxinaflash.com/&quot; rel=&quot;nofollow&quot;&gt;tutorial&lt;/a&gt;?]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Thank you for the info on setting up forward. I have done everything and it works fine. However the problem i have now is there is a 4-5 sec delay in the audio conversation. This is very difficult to talk with. When i place a call direct from one of my extensions there is almost zero delay. So it seems like it has to do with the forward. Any ideas? </p>
<p><i>[WM: Have you implemented the tips on Getting Rid of One-Way Audio in our <a href="http://knol.pbxinaflash.com/" rel="nofollow">tutorial</a>?]</i></p>
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		<title>
		By: Bob		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-2673</link>

		<dc:creator><![CDATA[Bob]]></dc:creator>
		<pubDate>Fri, 18 May 2007 05:46:24 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-2673</guid>

					<description><![CDATA[Thanks for the articles, you&#039;re advise is always extremely useful!  I&#039;m curious of others are having this issue; I&#039;ve a system when two ZAP channels and an IAX2 trunk to another system.  

I find that when I setup an extension to dial out to a cell phone as well as ring a SIP extension on my system the SIP extension rings only once and then the call is handed off completely to the trunk (either zip or iax2) going to the cell phone.  Watching the Asterisk console, I see what looks like Asterisk thinking the trunk has &quot;answered&quot; the call so it stops ringing any other extensions.  The call does get completed to the cell phone but there&#039;s no chance to answer the call on the SIP.  I&#039;ve tried this in both hunt and ring-all configs.

I&#039;m currently running Trixbox 2.2 but I&#039;ve had this problem through several versions of Trixbox/AAH.  Any ideas?

&lt;i&gt;[WM: I&#039;ve seen the same thing going all the way back to Asterisk@Home 1.5. It&#039;s definitely a bug, but no one seems to be able to find or fix it.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Thanks for the articles, you&#8217;re advise is always extremely useful!  I&#8217;m curious of others are having this issue; I&#8217;ve a system when two ZAP channels and an IAX2 trunk to another system.  </p>
<p>I find that when I setup an extension to dial out to a cell phone as well as ring a SIP extension on my system the SIP extension rings only once and then the call is handed off completely to the trunk (either zip or iax2) going to the cell phone.  Watching the Asterisk console, I see what looks like Asterisk thinking the trunk has "answered" the call so it stops ringing any other extensions.  The call does get completed to the cell phone but there&#8217;s no chance to answer the call on the SIP.  I&#8217;ve tried this in both hunt and ring-all configs.</p>
<p>I&#8217;m currently running Trixbox 2.2 but I&#8217;ve had this problem through several versions of Trixbox/AAH.  Any ideas?</p>
<p><i>[WM: I&#8217;ve seen the same thing going all the way back to Asterisk@Home 1.5. It&#8217;s definitely a bug, but no one seems to be able to find or fix it.]</i></p>
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		<title>
		By: Ice		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-2658</link>

		<dc:creator><![CDATA[Ice]]></dc:creator>
		<pubDate>Fri, 11 May 2007 06:16:15 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-2658</guid>

					<description><![CDATA[I cannot create a new user on StanaPhone.com. Sign up for an account. is not allowed. Exist any other way? Please let me know. Am so frustrated. Thanks.

&lt;i&gt;[WM: Stanaphone at least temporarily is not accepting new accounts.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>I cannot create a new user on StanaPhone.com. Sign up for an account. is not allowed. Exist any other way? Please let me know. Am so frustrated. Thanks.</p>
<p><i>[WM: Stanaphone at least temporarily is not accepting new accounts.]</i></p>
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		<title>
		By: Nick		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-2015</link>

		<dc:creator><![CDATA[Nick]]></dc:creator>
		<pubDate>Mon, 09 Oct 2006 23:03:06 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-2015</guid>

					<description><![CDATA[I&#039;m having trouble with this. My asterisk extensions just ring once, however it does make the call out to my cell phone.  Any thoughts?]]></description>
			<content:encoded><![CDATA[<p>I&#8217;m having trouble with this. My asterisk extensions just ring once, however it does make the call out to my cell phone.  Any thoughts?</p>
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		<title>
		By: jack		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-1675</link>

		<dc:creator><![CDATA[jack]]></dc:creator>
		<pubDate>Thu, 22 Jun 2006 01:56:07 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-1675</guid>

					<description><![CDATA[If you&#039;re having trouble setting up Stanaphone like I did check out http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+6#614Stanaphone it worked for me.]]></description>
			<content:encoded><![CDATA[<p>If you&#8217;re having trouble setting up Stanaphone like I did check out <a href="http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+6#614Stanaphone" rel="nofollow ugc">http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+6#614Stanaphone</a> it worked for me.</p>
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		<title>
		By: DanITman		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-1167</link>

		<dc:creator><![CDATA[DanITman]]></dc:creator>
		<pubDate>Mon, 13 Mar 2006 03:28:17 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-1167</guid>

					<description><![CDATA[How would you pass the original caller ID from the incoming call to show up on your cell phone?  This is a popular feature that Avaya systems offer.

&lt;i&gt;[WM: Great question, and you&#039;re right.  It was a must-have in our book. Read our &lt;a href=&quot;http://nerdvittles.com/index.php?p=115&quot;&gt;article&lt;/a&gt; on that very subject.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>How would you pass the original caller ID from the incoming call to show up on your cell phone?  This is a popular feature that Avaya systems offer.</p>
<p><i>[WM: Great question, and you&#8217;re right.  It was a must-have in our book. Read our <a href="http://nerdvittles.com/index.php?p=115">article</a> on that very subject.]</i></p>
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		<title>
		By: Bob Thompson		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-1122</link>

		<dc:creator><![CDATA[Bob Thompson]]></dc:creator>
		<pubDate>Fri, 03 Mar 2006 16:20:10 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-1122</guid>

					<description><![CDATA[First off if it wasn&#039;t for you and your articles I wouldn&#039;t have an Asterisk system up and running support WorldWide business operations. Thank you!

In a previous posting you stated that the Dial line for an extentsion had to changed from Sip/XX to Dial(local/XX@from-internal) where XX is the extentsion number for follow me calling to work. I have found this to be a true statement with the Asterisk@home version 2.5 that I am currently running. Was it an oversite that this information wasn&#039;t include in this article or have you found a way around this issue?]]></description>
			<content:encoded><![CDATA[<p>First off if it wasn&#8217;t for you and your articles I wouldn&#8217;t have an Asterisk system up and running support WorldWide business operations. Thank you!</p>
<p>In a previous posting you stated that the Dial line for an extentsion had to changed from Sip/XX to Dial(local/XX@from-internal) where XX is the extentsion number for follow me calling to work. I have found this to be a true statement with the Asterisk@home version 2.5 that I am currently running. Was it an oversite that this information wasn&#8217;t include in this article or have you found a way around this issue?</p>
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		<title>
		By: Aaron		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-1117</link>

		<dc:creator><![CDATA[Aaron]]></dc:creator>
		<pubDate>Thu, 02 Mar 2006 22:00:40 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-1117</guid>

					<description><![CDATA[Does it really need a ] added to make it work? This is how mine looks without changes. What should it look like?

exten =&gt; s,22,GotoIf($[$[${HuntMembers}] &gt;= 1]?30 )

&lt;i&gt;[WM: Yours looks correct. What version of Asterisk@Home?]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Does it really need a ] added to make it work? This is how mine looks without changes. What should it look like?</p>
<p>exten => s,22,GotoIf($[$[${HuntMembers}] >= 1]?30 )</p>
<p><i>[WM: Yours looks correct. What version of Asterisk@Home?]</i></p>
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		<title>
		By: Dave		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-1111</link>

		<dc:creator><![CDATA[Dave]]></dc:creator>
		<pubDate>Wed, 01 Mar 2006 19:28:11 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-1111</guid>

					<description><![CDATA[I love it. I already did this (two nights ago actually) but there are always so many little things in your articles.  I set up an extension and forwarded it to my cell phone, instead of simply typing the number in the right group like you did (duh) and you always come up with the best voip providers and freebies. 
Thanks again for the great articles.]]></description>
			<content:encoded><![CDATA[<p>I love it. I already did this (two nights ago actually) but there are always so many little things in your articles.  I set up an extension and forwarded it to my cell phone, instead of simply typing the number in the right group like you did (duh) and you always come up with the best voip providers and freebies.<br />
Thanks again for the great articles.</p>
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		<title>
		By: mike myers		</title>
		<link>https://nerdvittles.com/transparent-integration-of-mobile-phones-into-asterisk/comment-page-1/#comment-1110</link>

		<dc:creator><![CDATA[mike myers]]></dc:creator>
		<pubDate>Wed, 01 Mar 2006 18:57:10 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=118#comment-1110</guid>

					<description><![CDATA[Thanks for the great article.  I want asterisk to handle all my voicemail.  I can get my cell phone voicemail shut off, so it just rings, but if my phone battery runs out or I have my phone off (like when I am on an airplane), the cell carrier immediately answers the call with an intercept. My friends have similar issues.  

Do you know a way to get asterisk to do the right thing when the cell call is answered by an intercept or voicemail?

Thanks,
mike

&lt;i&gt;[WM: The problem is that Asterisk can only detect when a call is answered by a PSTN line or voicemail service so it has no way to distinguish between a person and a voicemail system. One possible solution is to require that some key be pressed when you answer a cellphone call. No key press = voicemail. I&#039;ll work on it a bit.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Thanks for the great article.  I want asterisk to handle all my voicemail.  I can get my cell phone voicemail shut off, so it just rings, but if my phone battery runs out or I have my phone off (like when I am on an airplane), the cell carrier immediately answers the call with an intercept. My friends have similar issues.  </p>
<p>Do you know a way to get asterisk to do the right thing when the cell call is answered by an intercept or voicemail?</p>
<p>Thanks,<br />
mike</p>
<p><i>[WM: The problem is that Asterisk can only detect when a call is answered by a PSTN line or voicemail service so it has no way to distinguish between a person and a voicemail system. One possible solution is to require that some key be pressed when you answer a cellphone call. No key press = voicemail. I&#8217;ll work on it a bit.]</i></p>
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