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	<title>i summon one kim &#187; VoIP</title>
	<atom:link href="http://kimmo.suominen.com/archives/tag/voip/feed/" rel="self" type="application/rss+xml" />
	<link>http://kimmo.suominen.com</link>
	<description>The website of Kimmo Suominen</description>
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		<item>
		<title>Bring back audio after Asterisk 1.6 upgrade</title>
		<link>http://kimmo.suominen.com/archives/2011/07/bring-back-audio-after-asterisk-1-6-upgrade/</link>
		<comments>http://kimmo.suominen.com/archives/2011/07/bring-back-audio-after-asterisk-1-6-upgrade/#comments</comments>
		<pubDate>Mon, 04 Jul 2011 19:03:00 +0000</pubDate>
		<dc:creator>Kimmo Suominen</dc:creator>
				<category><![CDATA[computers]]></category>
		<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://kimmo.suominen.com/?p=391</guid>
		<description><![CDATA[I upgraded to Asterisk 1.6 some time ago, but didn&#8217;t think anything was wrong until recently. Calls coming in from Callcentric didn&#8217;t work: I received no audio. Everything had been working fine with Asterisk 1.4. I don&#8217;t get many calls, so initially I dismissed this as a temporary problem. Calls from my other four carriers [...]]]></description>
			<content:encoded><![CDATA[<p>I upgraded to <a href="http://www.asterisk.org/">Asterisk</a> 1.6 some time ago, but didn&#8217;t think anything was wrong until recently. Calls coming in from <a href="http://www.callcentric.com/">Callcentric</a> didn&#8217;t work: I received no audio. Everything had been working fine with Asterisk 1.4. I don&#8217;t get many calls, so initially I dismissed this as a temporary problem. Calls from my other four carriers kept working fine.</p>

<p>After some research, I noticed the following settings <a href="http://www.callcentric.com/support/device/asterisk/1_6">suggested by Callcentric</a>:</p>

<blockquote>
<pre><code>session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas
</code></pre>
</blockquote>

<p>I&#8217;ve placed this in the <code>[general]</code> section of <code>sip.conf</code>, because calls from Callcentric arrive from multiple servers and the way Asterisk handles SRV records, only one of the servers ends up mapping into the per-carrier context at any given time. It doesn&#8217;t seem to have an adverse effect on calls from other carriers. (It is just turning off functionality new to 1.6, and setting some sensible defaults.)</p>]]></content:encoded>
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		</item>
		<item>
		<title>Ekiga on the intranet</title>
		<link>http://kimmo.suominen.com/archives/2009/06/ekiga-on-the-intranet/</link>
		<comments>http://kimmo.suominen.com/archives/2009/06/ekiga-on-the-intranet/#comments</comments>
		<pubDate>Sat, 13 Jun 2009 13:25:07 +0000</pubDate>
		<dc:creator>Kimmo Suominen</dc:creator>
				<category><![CDATA[software]]></category>
		<category><![CDATA[Ekiga]]></category>
		<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://kimmo.suominen.com/?p=305</guid>
		<description><![CDATA[Ekiga has a problem talking from behind a firewall, or maybe the problem is that both the firewall and Ekiga are trying to be too clever. I&#8217;m running siproxd on the firewall to transparently handle SIP connections. All the hardware phones and ATAs as well as X-Lite work well, but Ekiga fails to register. The [...]]]></description>
			<content:encoded><![CDATA[<p><a href="http://ekiga.org/">Ekiga</a> has a problem talking from behind a firewall, or maybe the problem is that both the firewall and Ekiga are trying to be too clever. I&#8217;m running <a href="http://siproxd.sourceforge.net/">siproxd</a> on the firewall to transparently handle SIP connections. All the hardware phones and ATAs as well as <a href="http://www.counterpath.com/x-lite.html">X-Lite</a> work well, but Ekiga fails to register.</p>

<p>The fix is controversial: I&#8217;m <a href="http://wiki.ekiga.org/index.php/Ekiga_behind_a_NAT_router">disabling STUN</a>.</p>

<blockquote>
  <p><code>gconftool-2 -s /apps/ekiga/general/nat/stun_server --type=string</code></p>
</blockquote>

<p>However, this means Ekiga won&#8217;t work from less &#8220;intelligent&#8221; networks anymore. Since I&#8217;m considering Ekiga for my laptop, this might be a problem. Maybe I should try X-Lite under <a href="http://www.winehq.org/">Wine</a>.</p>]]></content:encoded>
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		</item>
		<item>
		<title>Nokia E65 on Asterisk</title>
		<link>http://kimmo.suominen.com/archives/2008/03/nokia-e65-on-asterisk/</link>
		<comments>http://kimmo.suominen.com/archives/2008/03/nokia-e65-on-asterisk/#comments</comments>
		<pubDate>Sun, 23 Mar 2008 15:11:24 +0000</pubDate>
		<dc:creator>Kimmo Suominen</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://kimmo.suominen.com/?p=238</guid>
		<description><![CDATA[I reconfigured my Nokia E65 mobile phone to connect to my Asterisk PBX, so I can make and receive SIP calls instead of cellular calls when I&#8217;m at home or at the office. I already had the corresponding WLANs configured on the phone for web browsing, so I didn&#8217;t have to do that part now [...]]]></description>
			<content:encoded><![CDATA[<p>I reconfigured my Nokia E65 mobile phone to connect to my Asterisk PBX, so I can make and receive SIP calls instead of cellular calls when I&#8217;m at home or at the office.  I already had the corresponding <acronym title="Wireless Local Area Network">WLAN</acronym>s configured on the phone for web browsing, so I didn&#8217;t have to do that part now (and thus those settings are not documented here).<span id="more-238"></span></p>

<table>
<tr>
<td colspan="4"><strong>Tools > Settings > Connection</strong></td>
</tr>
<tr>
<td>Access Points</td>
<td colspan="3">You should already have the <acronym title="Wireless Local Area Network">WLAN</acronym> connections defined</td>
</tr>
<tr>
<td rowspan="21">SIP Settings</td>
<td colspan="3">Add one SIP Profile for each <acronym title="Wireless Local Area Network">WLAN</acronym> Access Point</td>
</tr>
<tr>
<td>Profile name:</td>
<td colspan="2"><em>Example SIP</em></td>
</tr>
<tr>
<td>Service profile:</td>
<td colspan="2">IETF</td>
</tr>
<tr>
<td>Default access point:</td>
<td colspan="2"><em>Example <acronym title="Wireless Local Area Network">WLAN</acronym></em></td>
</tr>
<tr>
<td>Public user name:</td>
<td colspan="2"><em>4321@sip.example.com</em></td>
</tr>
<tr>
<td>Use compression:</td>
<td colspan="2">No</td>
</tr>
<tr>
<td>Registration:</td>
<td colspan="2">Always on</td>
</tr>
<tr>
<td>Use security:</td>
<td colspan="2">No</td>
</tr>
<tr>
<td rowspan="7">Proxy server</td>
<td>Proxy server address:</td>
<td><em>sip.example.com</em></td>
</tr>
<tr>
<td>Realm:</td>
<td>asterisk</td>
</tr>
<tr>
<td>User name:</td>
<td><em>4321</em></td>
</tr>
<tr>
<td>Password:</td>
<td><em>****</em></td>
</tr>
<tr>
<td>Allow loose routing:</td>
<td>Yes</td>
</tr>
<tr>
<td>Transport type:</td>
<td>UDP</td>
</tr>
<tr>
<td>Port:</td>
<td>5060</td>
</tr>
<tr>
<td rowspan="6">Registrar server</td>
<td>Registrar serv. addr.:</td>
<td><em>sip.example.com</em></td>
</tr>
<tr>
<td>Realm:</td>
<td>None</td>
</tr>
<tr>
<td>User name:</td>
<td>None</td>
</tr>
<tr>
<td>Password:</td>
<td>None</td>
</tr>
<tr>
<td>Transport type:</td>
<td>UDP</td>
</tr>
<tr>
<td>Port:</td>
<td>5060</td>
</tr>
<tr>
<td rowspan="2">Internet tel. settings</td>
<td>Name:</td>
<td colspan="2"><em>Example Asterisk</em></td>
</tr>
<tr>
<td>SIP profiles:</td>
<td colspan="2">Select all</td>
</tr>
<tr>
<td colspan="4"><strong>Tools > Settings > Call</strong></td>
</tr>
<tr>
<td>Default call type:</td>
<td colspan="3">Internet</td>
</tr>
</table>

<p>The only way to change the settings in the SIP profiles seems to be to delete everything and re-enter the data.  If I try to change anything, the phone just gives an error message about being unable to save the data, because &#8220;all profiles in the same realm must use the same account information&#8221;.  I&#8217;m not quite sure why that is happening.  It could be because I&#8217;m using &#8220;None&#8221; as the realm in the registrar server settings.  It might be &#8220;conflicting&#8221; with the (unused) SIP profile provided by my cellular service provider.</p>

<p>References:</p>

<ul>
<li><a href="http://forum.voxilla.com/voip-wiki/using-nokia-e-series-phones-asterisk-24215.html">http://forum.voxilla.com/voip-wiki/using-nokia-e-series-phones-asterisk-24215.html</a></li>
</ul>]]></content:encoded>
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		</item>
		<item>
		<title>Tuning the phones</title>
		<link>http://kimmo.suominen.com/archives/2008/01/tuning-the-phones/</link>
		<comments>http://kimmo.suominen.com/archives/2008/01/tuning-the-phones/#comments</comments>
		<pubDate>Tue, 01 Jan 2008 16:33:49 +0000</pubDate>
		<dc:creator>Kimmo Suominen</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://kimmo.suominen.com/archives/2008/01/tuning-the-phones/</guid>
		<description><![CDATA[I spent some time with my parents about a month ago fine tuning the echo cancellation on our SIP phones. Today I got a chance to refresh my memory while connecting the phones of a friend to a CallWeaver PBX I had setup for him. Hopefully I have now collected in one place all the [...]]]></description>
			<content:encoded><![CDATA[<p>I spent some time with my parents about a month ago fine tuning the echo cancellation on our SIP phones.  Today I got a chance to refresh my memory while connecting the phones of a friend to a <a href="http://www.callweaver.org/">CallWeaver</a> PBX I had setup for him.  Hopefully I have now collected in one place all the settings that were changed from the factory defaults.  If you have a <a href="http://www.google.com/search?q=spa3102">Linksys SPA3102</a> phone adapter or a <a href="http://www.google.com/search?q=spa942">Linksys SPA942</a> phone, you may find some useful hints in here. <span id="more-235"></span></p>

<p>First navigate to the advanced voice administration pages on your device:</p>

<ul>
<li><strong>Linksys SPA3102:</strong> Admin Login &gt; Advanced &gt; Voice</li>
<li><strong>Linksys SPA942:</strong> Admin Login &gt; Advanced</li>
</ul>

<p>The following settings work well for me:</p>

<table>
<tbody>
<tr>
<th>Tab</th>
<th>Section</th>
<th>Setting</th>
<th>Value</th>
<th>Remarks</th>
</tr>
<tr>
<td>SIP</td>
<td>RTP Parameters</td>
<td>RTP Packet Size</td>
<td>0.020</td>
<td></td>
</tr>
<tr>
<td rowspan="2">Regional</td>
<td rowspan="2">Miscellaneous</td>
<td>Time Zone</td>
<td>GMT +02:00</td>
<td>Finland</td>
</tr>
<tr>
<td>Daylight Saving Time Rule</td>
<td colspan="2">start=3/-1/7/3;end=10/-1/7/4;save=1<br />
(European Union)</td>
</tr>
<tr>
<td rowspan="10">Regional
<small>(SPA3102)</small></td>
<td>Ring and Call Waiting Tone Spec</td>
<td>Ring Waveform</td>
<td>Sinusoid</td>
<td>Finland</td>
</tr>
<tr>
<td>Control Timer Values</td>
<td>Hook Flash Timer Min</td>
<td>.05</td>
<td></td>
</tr>
<tr>
<td>Miscellaneous</td>
<td>More Echo Suppression</td>
<td>yes</td>
<td></td>
</tr>
<tr>
<td rowspan="4">Miscellaneous
<small>(Siemens Gigaset 3000)</small></td>
<td>FXS Port Input Gain</td>
<td>16</td>
<td></td>
</tr>
<tr>
<td>FXS Port Output Gain</td>
<td>-15</td>
<td></td>
</tr>
<tr>
<td>Caller ID Method</td>
<td colspan="2">Bellcore (N.Amer, China)</td>
</tr>
<tr>
<td>Caller ID FSK Standard</td>
<td>bell 202</td>
<td></td>
</tr>
<tr>
<td rowspan="3">Miscellaneous
<small>(Siemens Gigaset E450)</small></td>
<td>FXS Port Input Gain</td>
<td>0</td>
<td></td>
</tr>
<tr>
<td>FXS Port Output Gain</td>
<td>-3</td>
<td></td>
</tr>
<tr>
<td>Caller ID Method</td>
<td colspan="2">DTMF (Finland, Sweden)</td>
</tr>
<tr>
<td rowspan="3">Phone
<small>(SPA942)</small></td>
<td rowspan="3">Audio Input Gain (dB)</td>
<td>Handset Input Gain</td>
<td>6</td>
<td></td>
</tr>
<tr>
<td>Headset Input Gain</td>
<td>0</td>
<td>Hello Direct Unaural</td>
</tr>
<tr>
<td>Acoustic Echo Canceller Enable</td>
<td>yes</td>
<td></td>
</tr>
<tr>
<td rowspan="3">Ext 1-4
<small>(SPA942)</small>
Line 1
<small>(SPA3102)</small></td>
<td rowspan="2">Network Settings</td>
<td>Network Jitter Level</td>
<td>low</td>
<td>Echo suppression</td>
</tr>
<tr>
<td>Jitter Buffer Adjustment</td>
<td>up and down</td>
<td>Echo suppression</td>
</tr>
<tr>
<td>Proxy and Registration</td>
<td>Register Expires</td>
<td>120</td>
<td>Notice network problems faster</td>
</tr>
<tr>
<td rowspan="2">User
<small>(SPA942)</small></td>
<td>Supplementary Services</td>
<td>Date Format</td>
<td>day/month</td>
<td></td>
</tr>
<tr>
<td>Audio Volume</td>
<td>Back Light Timer</td>
<td>30 s</td>
<td>Maximum timeout</td>
</tr>
</tbody></table>]]></content:encoded>
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		</item>
		<item>
		<title>Vonage horror stories</title>
		<link>http://kimmo.suominen.com/archives/2006/05/vonage-horror-stories/</link>
		<comments>http://kimmo.suominen.com/archives/2006/05/vonage-horror-stories/#comments</comments>
		<pubDate>Sun, 14 May 2006 17:04:46 +0000</pubDate>
		<dc:creator>Kimmo Suominen</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://kimmo.suominen.com/archives/2006/05/vonage-horror-stories/</guid>
		<description><![CDATA[I could have saved some time if I had read more user comments from the Google results for unlocking my Vonage-provided Cisco ATA-186 adapter. It appears Vonage doesn&#8217;t want to unlock your adapter even after you have stayed with them for ages, or even if you pay them for the unlocking service. I found this [...]]]></description>
			<content:encoded><![CDATA[<p>I could have saved some time if I had read more user comments from the Google results for unlocking my <a href="http://www.vonage.com/">Vonage</a>-provided Cisco ATA-186 adapter.  It appears Vonage doesn&#8217;t want to unlock your adapter even after you have stayed with them for ages, or even if you pay them for the unlocking service.</p>

<p>I found this quite surprising, as I&#8217;ve been a satisfied Vonage customer for many years.  I only discontinued the service several months after moving to Finland, when it was clear I was not using it enough to justify the cost.</p>

<p>Based on my experience today, though, it seems the horror stories are quite true.<span id="more-201"></span></p>

<p>I called the Vonage support number and explained that I&#8217;m no longer a customer but would like to use the adapter in my own network, and would need to have the adapter unlocked.  The support technician answered (more than once) that I could &#8220;simply unplug all cables from the adapter and it would not disturb the network.&#8221;  Obviously this did not answer my inquiry at all.</p>

<p>I wasn&#8217;t sure if the tech even understood my question (his English wasn&#8217;t very fluent), explained this to him, and asked to speak to a supervisor.  He offered to transfer the call to technical support instead of the supervisor, and I agreed to this.  It&#8217;s just that nobody ever picked up the call after that&#8230;</p>

<p>I guess all this nonsense was just tactics right from the beginning to get rid of me with my questions.  I would have preferred a direct answer like &#8220;sorry, we don&#8217;t provide unlocking for the adapters&#8221; or whatever the case is.  I wonder if that is against consumer laws in the USA, so they try to just avoid answering anything at all.  What other reason could there be for not being direct with customers?</p>

<p>Unfortunately my ATA-186 is of the later revision where the eeprom chip is soldered on the motherboard.  Even if I had eeprom programming hardware, it would take some careful soldering to retrieve the chip without damaging it (or the motherboard).</p>

<p>One more item for the next electronics recycling pickup&#8230;</p>]]></content:encoded>
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