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	<title>
	Comments on: SIP Proxies Make Asterisk Shine and Save You Money	</title>
	<atom:link href="https://nerdvittles.com/sip-proxies-make-asterisk-shine-and-save-you-money/feed/" rel="self" type="application/rss+xml" />
	<link>https://nerdvittles.com/sip-proxies-make-asterisk-shine-and-save-you-money/</link>
	<description>Ward Mundy&#039;s Technobabblelog</description>
	<lastBuildDate>Wed, 09 Dec 2015 13:04:28 +0000</lastBuildDate>
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	<item>
		<title>
		By: Chris		</title>
		<link>https://nerdvittles.com/sip-proxies-make-asterisk-shine-and-save-you-money/comment-page-1/#comment-13482</link>

		<dc:creator><![CDATA[Chris]]></dc:creator>
		<pubDate>Thu, 20 Jan 2011 15:16:20 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=208#comment-13482</guid>

					<description><![CDATA[You talk about the virtues of Google Talk. But does it have the encryption found in Skype?]]></description>
			<content:encoded><![CDATA[<p>You talk about the virtues of Google Talk. But does it have the encryption found in Skype?</p>
]]></content:encoded>
		
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		<title>
		By: Rafael Cortes		</title>
		<link>https://nerdvittles.com/sip-proxies-make-asterisk-shine-and-save-you-money/comment-page-1/#comment-9014</link>

		<dc:creator><![CDATA[Rafael Cortes]]></dc:creator>
		<pubDate>Wed, 04 Mar 2009 00:52:18 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=208#comment-9014</guid>

					<description><![CDATA[Adorah setup here is incomplete and I couldn&#039;t figure out the problem... but here&#039;s how to add IPKall as a Sip Trunk in FreePBX thanks to http://www.voip-info.org/wiki/view/IPKall 

Settings for add Trunk in freepbx  

1. click add sip trunk 
2. leave everything blank except the following items 

Outgoing Settings 

Trunk Name: ipkall 
Peer Details: 

context=from-pstn 
host=voiper.ipkall.com 
type=peer 

Incoming Settings 

***** USER Context from-trunk *** I changed this to be able to use it without changing a thing... I changed from-external as the USER Context to the number I want to use in the Phone Number in IPKall

USER Context: 98765

USER Details: 

allow=g729 
context=from-trunk 
disallow=all 
dtmfmode=rfc2833 
host=voiper.ipkall.com 
insecure=very 
type=peer 

Now in IPKall I would use Phone Number 98765 server address mypbx.dyndns.org
My sip address in this example is 98765@mypbx.dyndns.org

Thanks for everything!]]></description>
			<content:encoded><![CDATA[<p>Adorah setup here is incomplete and I couldn&#8217;t figure out the problem&#8230; but here&#8217;s how to add IPKall as a Sip Trunk in FreePBX thanks to <a href="http://www.voip-info.org/wiki/view/IPKall" rel="nofollow ugc">http://www.voip-info.org/wiki/view/IPKall</a> </p>
<p>Settings for add Trunk in freepbx  </p>
<p>1. click add sip trunk<br />
2. leave everything blank except the following items </p>
<p>Outgoing Settings </p>
<p>Trunk Name: ipkall<br />
Peer Details: </p>
<p>context=from-pstn<br />
host=voiper.ipkall.com<br />
type=peer </p>
<p>Incoming Settings </p>
<p>***** USER Context from-trunk *** I changed this to be able to use it without changing a thing&#8230; I changed from-external as the USER Context to the number I want to use in the Phone Number in IPKall</p>
<p>USER Context: 98765</p>
<p>USER Details: </p>
<p>allow=g729<br />
context=from-trunk<br />
disallow=all<br />
dtmfmode=rfc2833<br />
host=voiper.ipkall.com<br />
insecure=very<br />
type=peer </p>
<p>Now in IPKall I would use Phone Number 98765 server address mypbx.dyndns.org<br />
My sip address in this example is <a href="mailto:98765@mypbx.dyndns.org">98765@mypbx.dyndns.org</a></p>
<p>Thanks for everything!</p>
]]></content:encoded>
		
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		<item>
		<title>
		By: Solarion		</title>
		<link>https://nerdvittles.com/sip-proxies-make-asterisk-shine-and-save-you-money/comment-page-1/#comment-3204</link>

		<dc:creator><![CDATA[Solarion]]></dc:creator>
		<pubDate>Fri, 14 Mar 2008 18:06:07 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=208#comment-3204</guid>

					<description><![CDATA[Thanks Ward for burning your midnight oil and telling us of this feature. Yes we too did the trunk setup like adorah above said. You always help us keep our bells ringing, to a different tone.]]></description>
			<content:encoded><![CDATA[<p>Thanks Ward for burning your midnight oil and telling us of this feature. Yes we too did the trunk setup like adorah above said. You always help us keep our bells ringing, to a different tone.</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Michael Ruge		</title>
		<link>https://nerdvittles.com/sip-proxies-make-asterisk-shine-and-save-you-money/comment-page-1/#comment-3202</link>

		<dc:creator><![CDATA[Michael Ruge]]></dc:creator>
		<pubDate>Thu, 13 Mar 2008 15:37:02 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=208#comment-3202</guid>

					<description><![CDATA[For some reason, I have only been able to get one way audio with IPKall.  I have had this problem for years and have never been able to resolve the issue.  The correct ports are open on my router.  I can use other sip services.  The only one that wont work is IPKall.

&lt;i&gt;[WM: In your sip_custom.conf file, have you added the code below using the external ip and internal subnet for your network? Also be sure to read the new section in the article about Avoiding NAT Problems.]
[code]
externip=66.56.246.186
localnet=192.168.0.0/255.255.255.0
[/code]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>For some reason, I have only been able to get one way audio with IPKall.  I have had this problem for years and have never been able to resolve the issue.  The correct ports are open on my router.  I can use other sip services.  The only one that wont work is IPKall.</p>
<p><i>[WM: In your sip_custom.conf file, have you added the code below using the external ip and internal subnet for your network? Also be sure to read the new section in the article about Avoiding NAT Problems.]<br />
[code]<br />
externip=66.56.246.186<br />
localnet=192.168.0.0/255.255.255.0<br />
[/code]</i></p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: adorah		</title>
		<link>https://nerdvittles.com/sip-proxies-make-asterisk-shine-and-save-you-money/comment-page-1/#comment-3200</link>

		<dc:creator><![CDATA[adorah]]></dc:creator>
		<pubDate>Wed, 12 Mar 2008 23:47:01 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=208#comment-3200</guid>

					<description><![CDATA[You may configure IPKALL through FREEPBX by adding a SIP trunk.
simply fill in the incoming setting without registering:
USER CONTEXT: 588
type=peer
insecure=very
host=voiper.ipkall.com
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=g729 (or ulaw)
And that&#039;s it. no need for user name or password..
One more thing: Dynamic DNS can also be obtained from www.no-ip.com,  for free.]]></description>
			<content:encoded><![CDATA[<p>You may configure IPKALL through FREEPBX by adding a SIP trunk.<br />
simply fill in the incoming setting without registering:<br />
USER CONTEXT: 588<br />
type=peer<br />
insecure=very<br />
host=voiper.ipkall.com<br />
dtmfmode=rfc2833<br />
disallow=all<br />
context=from-trunk<br />
allow=g729 (or ulaw)<br />
And that&#8217;s it. no need for user name or password..<br />
One more thing: Dynamic DNS can also be obtained from <a href="http://www.no-ip.com" rel="nofollow ugc">http://www.no-ip.com</a>,  for free.</p>
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