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	<title>
	Comments on: Adding SIP URI Dialing to Asterisk for Free Worldwide Calling	</title>
	<atom:link href="https://nerdvittles.com/adding-sip-uri-dialing-to-asterisk-for-free-worldwide-calling/feed/" rel="self" type="application/rss+xml" />
	<link>https://nerdvittles.com/adding-sip-uri-dialing-to-asterisk-for-free-worldwide-calling/</link>
	<description>Ward Mundy&#039;s Technobabblelog</description>
	<lastBuildDate>Tue, 20 Jul 2021 14:04:26 +0000</lastBuildDate>
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		<title>
		By: ceskaiser		</title>
		<link>https://nerdvittles.com/adding-sip-uri-dialing-to-asterisk-for-free-worldwide-calling/comment-page-1/#comment-178333</link>

		<dc:creator><![CDATA[ceskaiser]]></dc:creator>
		<pubDate>Wed, 08 May 2019 12:50:33 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=27781#comment-178333</guid>

					<description><![CDATA[Hello!

Great, fantastic work on Incredible PBX! I cannot thank you enough!

I have installed the URI mod on May 7th, and coincidentally right after, my SPA2102 cannot do outgoing calls, not even to local extensions. It looks as if I am missing a setting that was not needed before, as asterisk sees the call requests as non registered. Error below:
&quot;Rejecting unknown SIP connection from [PBXIPAddress]&quot;
PBXIPAddress is my pbx ip address, which I did not want published.

What is curious, is that my IP 1535 and Snom 320 phones are working fine, it only affected the SPA2102.

Any hints of what I would need to change in the SPA2102?

&lt;i&gt;[WM: This is difficult to diagnose in a blog comment. Open a thread on the PIAF Forum and post the output from the Asterisk CLI when you make a call from your SPA2102, and we&#039;ll help you sort it out.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Hello!</p>
<p>Great, fantastic work on Incredible PBX! I cannot thank you enough!</p>
<p>I have installed the URI mod on May 7th, and coincidentally right after, my SPA2102 cannot do outgoing calls, not even to local extensions. It looks as if I am missing a setting that was not needed before, as asterisk sees the call requests as non registered. Error below:<br />
"Rejecting unknown SIP connection from [PBXIPAddress]"<br />
PBXIPAddress is my pbx ip address, which I did not want published.</p>
<p>What is curious, is that my IP 1535 and Snom 320 phones are working fine, it only affected the SPA2102.</p>
<p>Any hints of what I would need to change in the SPA2102?</p>
<p><i>[WM: This is difficult to diagnose in a blog comment. Open a thread on the PIAF Forum and post the output from the Asterisk CLI when you make a call from your SPA2102, and we&#8217;ll help you sort it out.]</i></p>
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		<item>
		<title>
		By: Wilbert		</title>
		<link>https://nerdvittles.com/adding-sip-uri-dialing-to-asterisk-for-free-worldwide-calling/comment-page-1/#comment-177811</link>

		<dc:creator><![CDATA[Wilbert]]></dc:creator>
		<pubDate>Wed, 13 Feb 2019 08:39:58 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=27781#comment-177811</guid>

					<description><![CDATA[Found the issue. And btw, your Gotoif conditions had &quot;)&quot; missing.

[ext-local-custom]
exten =&#062; _.,1,Set(MyDomain=${CUT(SIPCALLID,@,2)})
exten =&#062; _.,n,NoOp(SIPDOMAIN: ${SIPDOMAIN})
exten =&#062; _.,n,GotoIf($[&quot;foo${MyDomain}&quot; != &quot;foo&quot;]?FoundDomain:OutAllRoutes)
exten =&#062; _.,n,Set(MyDomain=${DB(MyDomain/FQDN)})
exten =&#062; _.,n(FoundDomain),GotoIf($[&quot;${MyDomain}&quot; = &quot;${SIPDOMAIN}&quot;]?OutAllRoutes)
exten =&#062; _.,n,Macro(uridial,${EXTEN}@${SIPDOMAIN})
exten =&#062; _.,n(OutAllRoutes),Goto(outbound-allroutes,${EXTEN},1)

When a CallForwarding is executed MyDomain variable will be empty so will execute macro-uridial.

Will test during these days but working now.

&lt;i&gt;[WM:  Right you are. Good eyes! Code has been fixed, and command sequence can be run again to load the patched version. Many thanks.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Found the issue. And btw, your Gotoif conditions had ")" missing.</p>
<p>[ext-local-custom]<br />
exten =&gt; _.,1,Set(MyDomain=${CUT(SIPCALLID,@,2)})<br />
exten =&gt; _.,n,NoOp(SIPDOMAIN: ${SIPDOMAIN})<br />
exten =&gt; _.,n,GotoIf($["foo${MyDomain}" != "foo"]?FoundDomain:OutAllRoutes)<br />
exten =&gt; _.,n,Set(MyDomain=${DB(MyDomain/FQDN)})<br />
exten =&gt; _.,n(FoundDomain),GotoIf($["${MyDomain}" = "${SIPDOMAIN}"]?OutAllRoutes)<br />
exten =&gt; _.,n,Macro(uridial,${EXTEN}@${SIPDOMAIN})<br />
exten =&gt; _.,n(OutAllRoutes),Goto(outbound-allroutes,${EXTEN},1)</p>
<p>When a CallForwarding is executed MyDomain variable will be empty so will execute macro-uridial.</p>
<p>Will test during these days but working now.</p>
<p><i>[WM:  Right you are. Good eyes! Code has been fixed, and command sequence can be run again to load the patched version. Many thanks.]</i></p>
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		<item>
		<title>
		By: Wilbert Zapata		</title>
		<link>https://nerdvittles.com/adding-sip-uri-dialing-to-asterisk-for-free-worldwide-calling/comment-page-1/#comment-177809</link>

		<dc:creator><![CDATA[Wilbert Zapata]]></dc:creator>
		<pubDate>Tue, 12 Feb 2019 18:50:06 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=27781#comment-177809</guid>

					<description><![CDATA[Hi. It seems it&#039;s interfering with Follow me functionality when forwarding to an external number. It thinks is a uri dialing.

&lt;i&gt;[WM: Have you tried adding an entry to the Asterisk database from the Asterisk CLI using syntax like the following: &lt;strong&gt;database put MyDomain FQDN 10.0.0.11&lt;/strong&gt; or &lt;strong&gt;database put MyDomain FQDN sip.me.com&lt;/strong&gt; where &lt;strong&gt;10.0.0.11&lt;/strong&gt; or &lt;strong&gt;sip.me.com&lt;/strong&gt; matches the &lt;strong&gt;SIPDOMAIN&lt;/strong&gt; entry shown on the Asterisk CLI when you forward a test call?]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Hi. It seems it&#8217;s interfering with Follow me functionality when forwarding to an external number. It thinks is a uri dialing.</p>
<p><i>[WM: Have you tried adding an entry to the Asterisk database from the Asterisk CLI using syntax like the following: <strong>database put MyDomain FQDN 10.0.0.11</strong> or <strong>database put MyDomain FQDN sip.me.com</strong> where <strong>10.0.0.11</strong> or <strong>sip.me.com</strong> matches the <strong>SIPDOMAIN</strong> entry shown on the Asterisk CLI when you forward a test call?]</i></p>
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