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	<title>djerk.nl &#187; VoIP</title>
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	<link>http://www.djerk.nl/wordpress</link>
	<description>Anything related to Djerk Geurts may be found here (either locally or linked)</description>
	<lastBuildDate>Mon, 12 Nov 2012 12:44:27 +0000</lastBuildDate>
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		<title>Cisco Voice-VLAN (VVLAN) inconsistencies</title>
		<link>http://www.djerk.nl/wordpress/2012/cisco-voice-vlan-vvlan-inconsistencies</link>
		<comments>http://www.djerk.nl/wordpress/2012/cisco-voice-vlan-vvlan-inconsistencies#comments</comments>
		<pubDate>Mon, 12 Nov 2012 12:41:51 +0000</pubDate>
		<dc:creator>Djerk</dc:creator>
				<category><![CDATA[Networking]]></category>
		<category><![CDATA[VoIP]]></category>
		<category><![CDATA[Work]]></category>

		<guid isPermaLink="false">http://www.djerk.nl/wordpress/?p=567</guid>
		<description><![CDATA[First off I&#8217;d like to say that this is just a minor issue, more relating for routers versus switch, I&#8217;m still a lot happier at how Cisco implements config and features as opposed to most if not all of their competitors&#8230; At a customer I&#8217;ve recently had to commit a grave operational sin; to connect [...]]]></description>
				<content:encoded><![CDATA[<p>First off I&#8217;d like to say that this is just a minor issue, more relating for routers versus switch, I&#8217;m still a lot happier at how Cisco implements config and features as opposed to most if not all of their competitors&#8230;</p>
<p>At a customer I&#8217;ve recently had to commit a grave operational sin; to connect a small switch at the end of a floor patch. These things are normally operational nightmares as they have a tendency to quickly bring an entire LAN environment down to its knees when such a &#8216;switch&#8217; is connected to the network twice. Always by accident but having management kick you for something someone else did is not anyone&#8217;s idea of fun. I won&#8217;t go into the underlying principles here as I&#8217;m assuming most who frequent my blog will know about broadcast storms, their causes and the tools and solutions available to mitigate the risks.</p>
<p>Our justification to operations was that we wanted a few more local LAN ports to test VoIP devices on than we had available through floor patches. As such I reasoned with Operations that this was a calculated choice to segregate our testing from the rest of the LAN but still make it as realistic as possible. Using the means available meant that I had to make do with a Cisco 1801. Single routed and 8 switched interfaces. Think of it as a router with one Ethernet interface and an 8 port HWIC-ESW nailed to it. Didn&#8217;t need the ATM or WiFi it has.</p>
<p>So I set out, disabling IP routing, admin down all non-Ethernet ports. set up the vlan database -old style, remember?-; I did not want this baby to participate in VTP, in fact I don&#8217;t think it even can! It&#8217;s limited to 8 vlans. Pulled two cables to it. One switched port as trunked with some data and voice vlans and configured the routed interface for management access.</p>
<p>All sweet and dandy, tested the BPDU-guard functionality prior to installation by connecting an access-port to the LAN. Clunk! it went down as desired, result I thought&#8230; Then when installing the LAN wouldn&#8217;t bring up the LAN port. Doh! I&#8217;d missed that the 1801 doesn&#8217;t send BPDU&#8217;s until a VLAN becomes active. I&#8217;d checked if spanning-tree was operational, and it wasn&#8217;t until I brought an interface up. So I disabled STP for all vlans in the VLAN database. Now my laptop received an IP address and the data VLANs all worked.</p>
<p>So, time to connect a Mitel phone. No dice, it received it&#8217;s first DHCP response with VLAn information, then it would just sit ennuncing it was waiting for a DHCP response. Dang, I&#8217;d configured the voice vlan so why did the switch not detect the phone, enable trunking so that the phone could send it&#8217;s DHCP request on the voice VLAN?</p>
<p>It was only when I started reading up on HWIC-ESW voice-VLAN config I noticed that Cisco hasn&#8217;t implemented the auto enable of dot1q trunking when a phone is detected&#8230; The solution is to add two lines of code; &#8220;switchport truck native vlan xyz&#8221; and &#8220;switchport mode trunk&#8221;. The crux is that this platform is at heart a router, not a native switch&#8230;</p>
<p><a title="Cisco EtherSwitch 4- and 9-Port High-Speed WAN Interface Cards" href="http://www.cisco.com/en/US/prod/collateral/routers/ps5853/product_data_sheet0900aecd8016bf0b.html" target="_blank">Cisco documentation</a></p>
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		<item>
		<title>VoIP to Skype bridge</title>
		<link>http://www.djerk.nl/wordpress/2007/voip-to-skype-bridge</link>
		<comments>http://www.djerk.nl/wordpress/2007/voip-to-skype-bridge#comments</comments>
		<pubDate>Sat, 14 Apr 2007 08:08:52 +0000</pubDate>
		<dc:creator>Djerk</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://www.djerk.nl/wordpress/wordpress/2007/voip-to-skype-bridge/</guid>
		<description><![CDATA[For me SIP is the best option with most countries that I call being free and the UK being equal or cheaper than my current land line carrier. However my family in the UK has a better deal with Skype when the majority of the calls are national (1.4p per min and 4.5p per call). [...]]]></description>
				<content:encoded><![CDATA[<p>For me SIP is the best option with most countries that I call being free and the UK being equal or cheaper than my current <a title="External link to Bel 1649" href="http://www.bel1649.nl" target="_blank">land line carrier</a>. However my family in the UK has a better deal with Skype when the majority of the calls are national (1.4p per min and 4.5p per call). So if they have a stand alone Skype phone and I have SIP phones, should we not be able to call each other for free?</p>
<p>My options are:</p>
<ol>
<li><a title="External link to rsdevs.com PSGw" href="http://www.rsdevs.com/psgw.shtml" target="_blank">PSGw</a> requires an additional pc and single Skype account per pc<a title="External link to rsdevs.com PSGw" href="http://www.rsdevs.com/psgw.shtml" target="_blank"><br />
</a></li>
<li><a title="External link to chanskype.com" href="http://www.chanskype.com/" target="_blank">Skype Asterisk channel</a> (chan_skype) Can only run on Asterisk but can run multiple instances of Skype through vnc</li>
</ol>
<p>Not for me:</p>
<ol>
<li>&#8216;<a title="External link to NCH.au" href="http://www.nch.com.au/skypetosip/index.html" target="_blank">Uplink</a> Skype to SIP Adapter&#8217; (windows required)</li>
<li><a title="External link to CooSIP.com" href="http://www.coosip.com" target="_blank">CooSIP</a>, no idea about price and it&#8217;s not available yet</li>
<li><a title="External link to skip2pbx.com" href="http://www.skip2pbx.com" target="_blank">Skip2PBX</a>, way too expensive</li>
<li><a title="External link to Pika technologies" href="http://www.pikatechnologies.com" target="_blank">Pika Connect for Skype</a>, way too expensive hardware &amp; licenses [2009-01-07 no more references to Skype on their site]</li>
</ol>
<p>chan_skype it will be&#8230;</p>
<blockquote><p>Edit (23/05/2007): Well it&#8217;s end of May and I haven&#8217;t been able to spend more time on this yet, if you want to do this on Asterisk then no problem but on AsteriskNOW it&#8217;s not that easy.</p></blockquote>
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		<item>
		<title>Voipbuster free calls</title>
		<link>http://www.djerk.nl/wordpress/2007/voipbuster-free-calls</link>
		<comments>http://www.djerk.nl/wordpress/2007/voipbuster-free-calls#comments</comments>
		<pubDate>Fri, 13 Apr 2007 21:16:18 +0000</pubDate>
		<dc:creator>Djerk</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://www.djerk.nl/wordpress/wordpress/2007/voipbuster-free-calls/</guid>
		<description><![CDATA[Voipbuster is sporting free calls to quite a number of countries, sadly the UK isn&#8217;t one of them but at 1cpm I&#8217;m not complaining. However it&#8217;s good to realise one thing: Voipbuster limits free call duration to one hour. After this hour the remote end get&#8217;s it&#8217;s call dropped while the local end is not [...]]]></description>
				<content:encoded><![CDATA[<p>Voipbuster is sporting free calls to quite a number of countries, sadly the UK isn&#8217;t one of them but at 1cpm I&#8217;m not complaining. However it&#8217;s good to realise one thing:</p>
<p>Voipbuster limits free call duration to one hour. After this hour the remote end get&#8217;s it&#8217;s call dropped while the local end is not notified. I presume this is to prevent automatic (re)dialing to these countries. One could circumvent this by static noise detection but that&#8217;s a little too advanced for most script kiddies.</p>
<p>Not an issue for me as I&#8217;ll just dial again when the remote end stops talking to me&#8230; <img src='http://www.djerk.nl/wordpress/wp-includes/images/smilies/icon_wink.gif' alt=';)' class='wp-smiley' /> </p>
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		<item>
		<title>AsteriskNOW &#8211; Install 1.4 Beta 2</title>
		<link>http://www.djerk.nl/wordpress/2007/asterisknow-install-14-beta-2</link>
		<comments>http://www.djerk.nl/wordpress/2007/asterisknow-install-14-beta-2#comments</comments>
		<pubDate>Fri, 13 Apr 2007 10:49:17 +0000</pubDate>
		<dc:creator>Djerk</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://www.djerk.nl/wordpress/wordpress/2007/asterisknow-install-14-beta-2/</guid>
		<description><![CDATA[After playing with the idea for a long time I decided it was time to implement a VoIP PBX. Reading up on the subject I found that I had 3 options which appealed to me: Asterisk (*) on Debian, AsteriskNOW (*NOW) or Trixbox. I opted for *NOW as it&#8217;s stripped bare and I hope I [...]]]></description>
				<content:encoded><![CDATA[<p>After playing with the idea for a long time I decided it was time to implement a VoIP PBX. Reading up on the subject I found that I had 3 options which appealed to me: <a href="http://www.asterisk.org" title="External link to Asterisk.org" target="_blank">Asterisk</a> (*) on Debian, <a href="http://www.asterisknow.org" title="External link to Asterisknow.org" target="_blank">AsteriskNOW</a> (*NOW) or <a href="http://www.trixbox.org" title="External link to Trixbox" target="_blank">Trixbox</a>. I opted for *NOW as it&#8217;s stripped bare and I hope I can add features as and when I need them. So far it&#8217;s gone well but I do have some additions I still need.</p>
<p>Installation of *NOW beta 4 (1.4.2) went fast on an Intel ISP2150 I had available, took about 30 minutes in total, including base config. Next I had to test it so I found a good free SIP soft-client in <a href="http://www.counterpath.com/index.php?menu=Products&amp;smenu=xlite" title="External link to X-lite 3.0" target="_blank">X-lite 3.0</a>. And off I was phoning from laptop to laptop, surprisingly I had no issues with my WLAN.</p>
<p>The next move was to add two SIP accounts, <a href="http://www.voipbuster.com" title="External link to Voipbuster" target="_blank">Voipbuster</a> and <a href="http://www.xeloq.com/" title="External link to Xeloq" target="_blank">Xeloq</a>. Both offer free national calls in NL additionally Voipbuster offers a range of free international calls while Xeloq has cheaper national mobile rates. Dialling from softphone to national landlines worked right away and the people I called didn&#8217;t even notice I wasn&#8217;t using my regular phone or line.</p>
<p>My server is now happily humming away in the datacenter and I have two 7960&#8242;s connected and registered across NAT, all is well so far. I&#8217;ve given some family members accounts and I&#8217;ve had a 2 hour conversation with my brother last weekend. We both have DSL and were quite happy with the quality of both the conversation and the connection. <img src='http://www.djerk.nl/wordpress/wp-includes/images/smilies/icon_smile.gif' alt=':)' class='wp-smiley' /> </p>
<p>Next my plan is to migrate my current landline number to a SIP carrier, not sure which one yet, and try to implement a SIP to Skype bridge. Further actions on my to do list are:</p>
<ul>
<li>Cancel my land line and save at least 18,50 Euro per month</li>
<li>Get a (SIP) Siemens S450 IP dect phone to replace my current old Philips (pstn) dect</li>
<li>Implement a phone book for the Cisco 7960 phones</li>
<li>Implement some services for the Cisco 7960 phones</li>
<li>Get a GSM SIP gateway and a mobile contract for backup/emergency calls and cost savings to national mobile numbers</li>
<li>Implement a registry of calls (CDR?) so I can tell whether my bills will be correct</li>
<li>Offer the SIP service to family members, using data from the previous point</li>
</ul>
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