<?xml version="1.0" encoding="UTF-8"?><rss version="2.0"
	xmlns:content="http://purl.org/rss/1.0/modules/content/"
	xmlns:dc="http://purl.org/dc/elements/1.1/"
	xmlns:atom="http://www.w3.org/2005/Atom"
	xmlns:sy="http://purl.org/rss/1.0/modules/syndication/"
		>
<channel>
	<title>Comments on: Configure Cisco IP Phones with Asterisk</title>
	<atom:link href="http://www.minded.ca/2009-12-16/configure-cisco-ip-phones-with-asterisk/feed/" rel="self" type="application/rss+xml" />
	<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/</link>
	<description>custom systems and software development group</description>
	<lastBuildDate>Mon, 30 Jan 2012 18:11:40 +0000</lastBuildDate>
	<sy:updatePeriod>hourly</sy:updatePeriod>
	<sy:updateFrequency>1</sy:updateFrequency>
	<generator>http://wordpress.org/?v=3.2.1</generator>
	<item>
		<title>By: JW</title>
		<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/#comment-418</link>
		<dc:creator>JW</dc:creator>
		<pubDate>Mon, 30 Jan 2012 18:11:40 +0000</pubDate>
		<guid isPermaLink="false">http://minded.ca/?p=52#comment-418</guid>
		<description>Your website has saved me hours of head-desking trying to get a 7940 registered with our Asterisk-based environment. Many thanks for the pointer of enabling NAT - it was all that was required in the end to get a fully-functioning SIP-version 7940 up and running!

@Andrzej: try enabling NAT with nat_enable: 1 and nat_address as your provider&#039;s STUN server. If that doesn&#039;t work, try enabling nat_received_processing to 1 instead of going through the nat_address. I managed to get ours to find the public IP address simply by setting nat_enable to 1.</description>
		<content:encoded><![CDATA[<p>Your website has saved me hours of head-desking trying to get a 7940 registered with our Asterisk-based environment. Many thanks for the pointer of enabling NAT &#8211; it was all that was required in the end to get a fully-functioning SIP-version 7940 up and running!</p>
<p>@Andrzej: try enabling NAT with nat_enable: 1 and nat_address as your provider&#8217;s STUN server. If that doesn&#8217;t work, try enabling nat_received_processing to 1 instead of going through the nat_address. I managed to get ours to find the public IP address simply by setting nat_enable to 1.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Andrzej Michalski</title>
		<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/#comment-379</link>
		<dc:creator>Andrzej Michalski</dc:creator>
		<pubDate>Thu, 19 Jan 2012 23:41:26 +0000</pubDate>
		<guid isPermaLink="false">http://minded.ca/?p=52#comment-379</guid>
		<description>I&#039;m in a bit of a quandary. I&#039;ve got a bunch of 7941&#039;s which I had running with an internal Asterisk PBX on our local LAN for years. We&#039;ve just moved to a new office and I decided to outsource the PBX bit to a company who also use Asterisk. They&#039;ve had an engineer in for two days so far and have now told me that the Cisco 7941&#039;s aren&#039;t compatible and that I&#039;ll need to buy something like a Polycom. That was bad news to me as we&#039;ve paid good money for the 7941&#039;s and my staff are really familiar with them so don&#039;t really want to change. So I asked what the issue was. Their response was along the lines of:
As we&#039;re running a private LAN we need to NAT out to their PBX on a public address. This we can do, and can successfully dial out. However incoming calls can&#039;t be routed back to the handset because due to slight RFC incompatibilities the Asterisk PBX has the private IP address rather than the public IP address...

Does this make sense? Some of it definitely does but from what I&#039;ve read I get the impression that what we&#039;re asking for can be achieved relatively simply with these handsets..

Any advice would be VERY welcome - the staff who need the phones working are beginning to give me menacing looks!!

BTW - I know one easy option would be to install the PBX in the office, but I&#039;m on a mission to try and reduce the number of servers that I have to look after so I&#039;d really like to outsource the hosting of this one....</description>
		<content:encoded><![CDATA[<p>I&#8217;m in a bit of a quandary. I&#8217;ve got a bunch of 7941&#8242;s which I had running with an internal Asterisk PBX on our local LAN for years. We&#8217;ve just moved to a new office and I decided to outsource the PBX bit to a company who also use Asterisk. They&#8217;ve had an engineer in for two days so far and have now told me that the Cisco 7941&#8242;s aren&#8217;t compatible and that I&#8217;ll need to buy something like a Polycom. That was bad news to me as we&#8217;ve paid good money for the 7941&#8242;s and my staff are really familiar with them so don&#8217;t really want to change. So I asked what the issue was. Their response was along the lines of:<br />
As we&#8217;re running a private LAN we need to NAT out to their PBX on a public address. This we can do, and can successfully dial out. However incoming calls can&#8217;t be routed back to the handset because due to slight RFC incompatibilities the Asterisk PBX has the private IP address rather than the public IP address&#8230;</p>
<p>Does this make sense? Some of it definitely does but from what I&#8217;ve read I get the impression that what we&#8217;re asking for can be achieved relatively simply with these handsets..</p>
<p>Any advice would be VERY welcome &#8211; the staff who need the phones working are beginning to give me menacing looks!!</p>
<p>BTW &#8211; I know one easy option would be to install the PBX in the office, but I&#8217;m on a mission to try and reduce the number of servers that I have to look after so I&#8217;d really like to outsource the hosting of this one&#8230;.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Asterisk + Cisco IP phones 7940G and 7911G &#124; Reza Samimi&#039;s Web Page</title>
		<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/#comment-369</link>
		<dc:creator>Asterisk + Cisco IP phones 7940G and 7911G &#124; Reza Samimi&#039;s Web Page</dc:creator>
		<pubDate>Mon, 16 Jan 2012 06:30:32 +0000</pubDate>
		<guid isPermaLink="false">http://minded.ca/?p=52#comment-369</guid>
		<description>[...] Configure Cisco IP Phones with Asterisk [...]</description>
		<content:encoded><![CDATA[<p>[...] Configure Cisco IP Phones with Asterisk [...]</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Cisco IP Phones 79XX with Asterisk &#124; Reza Samimi&#039;s Web Page</title>
		<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/#comment-352</link>
		<dc:creator>Cisco IP Phones 79XX with Asterisk &#124; Reza Samimi&#039;s Web Page</dc:creator>
		<pubDate>Sat, 07 Jan 2012 01:31:30 +0000</pubDate>
		<guid isPermaLink="false">http://minded.ca/?p=52#comment-352</guid>
		<description>[...] Configure Cisco IP Phones with Asterisk [...]</description>
		<content:encoded><![CDATA[<p>[...] Configure Cisco IP Phones with Asterisk [...]</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Martin Politick</title>
		<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/#comment-349</link>
		<dc:creator>Martin Politick</dc:creator>
		<pubDate>Sat, 31 Dec 2011 21:06:23 +0000</pubDate>
		<guid isPermaLink="false">http://minded.ca/?p=52#comment-349</guid>
		<description>Fixed,
The default templates in sip.conf are missing a non-natted definition. Need to add:

nat=no

into my phone definition to make it work.
Martin Politick</description>
		<content:encoded><![CDATA[<p>Fixed,<br />
The default templates in sip.conf are missing a non-natted definition. Need to add:</p>
<p>nat=no</p>
<p>into my phone definition to make it work.<br />
Martin Politick</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Martin Politick</title>
		<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/#comment-348</link>
		<dc:creator>Martin Politick</dc:creator>
		<pubDate>Fri, 30 Dec 2011 23:27:35 +0000</pubDate>
		<guid isPermaLink="false">http://minded.ca/?p=52#comment-348</guid>
		<description>Hi, I&#039;m trying to change PBX from SipXecs to Asterisk
my Cisco 7960 works fine with SipXecs but I can&#039;t get them to authenticate to Asterisk.  I&#039;ve even upgraded the firmware to 8.12.

I tried &quot;asterisk -vvvvvvvvvvr&quot; and &quot;core set debug 7&quot;
but when the phone tries to connect I don&#039;t see anything on CLI.
If I break my SIP003094C2B2B6.cnf and place a bad &quot;line1_name&quot;, then I see an error message in CLI 
[Dec 30 18:18:59] NOTICE[1179]: chan_sip.c:24431 handle_request_register: Registration from &#039;&#039; failed for &#039;192.168.11.100:50878&#039; - No matching peer found

I&#039;ve looked in /var/log/asterisk/ and nothing there in messages
netstat -an does show something listening on udp port 5060, but one
thing that is strange is the &quot;sip show peers&quot;
Name/username Host          Dyn Forcerport ACL Port     Status
famille       (Unspecified) D   N   A  0         Unmonitored
kasandra      (Unspecified) D   N   A  0        Unmonitored

The PORT IS 0 ???
That does not sound right.


I&#039;m running Ubuntu 11.04 with Asterisk 1.8 installed using the package manager (i.e. the binary version, I didn&#039;t compiled).

Can you help?
Thanks in advance,
Martin Politick.</description>
		<content:encoded><![CDATA[<p>Hi, I&#8217;m trying to change PBX from SipXecs to Asterisk<br />
my Cisco 7960 works fine with SipXecs but I can&#8217;t get them to authenticate to Asterisk.  I&#8217;ve even upgraded the firmware to 8.12.</p>
<p>I tried &#8220;asterisk -vvvvvvvvvvr&#8221; and &#8220;core set debug 7&#8243;<br />
but when the phone tries to connect I don&#8217;t see anything on CLI.<br />
If I break my SIP003094C2B2B6.cnf and place a bad &#8220;line1_name&#8221;, then I see an error message in CLI<br />
[Dec 30 18:18:59] NOTICE[1179]: chan_sip.c:24431 handle_request_register: Registration from &#8221; failed for &#8217;192.168.11.100:50878&#8242; &#8211; No matching peer found</p>
<p>I&#8217;ve looked in /var/log/asterisk/ and nothing there in messages<br />
netstat -an does show something listening on udp port 5060, but one<br />
thing that is strange is the &#8220;sip show peers&#8221;<br />
Name/username Host          Dyn Forcerport ACL Port     Status<br />
famille       (Unspecified) D   N   A  0         Unmonitored<br />
kasandra      (Unspecified) D   N   A  0        Unmonitored</p>
<p>The PORT IS 0 ???<br />
That does not sound right.</p>
<p>I&#8217;m running Ubuntu 11.04 with Asterisk 1.8 installed using the package manager (i.e. the binary version, I didn&#8217;t compiled).</p>
<p>Can you help?<br />
Thanks in advance,<br />
Martin Politick.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: g1smd</title>
		<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/#comment-342</link>
		<dc:creator>g1smd</dc:creator>
		<pubDate>Fri, 09 Dec 2011 23:48:38 +0000</pubDate>
		<guid isPermaLink="false">http://minded.ca/?p=52#comment-342</guid>
		<description>Anyone trying to make sense of their UK dialplan documentation should read the page at: http://www.aa-asterisk.org.uk/index.php/Errors_in_Cisco_UK_Numbering_Plan_Documents

If you want details of UK local dialling patterns within each area code, see the page at: http://www.aa-asterisk.org.uk/index.php/Regular_Expressions_for_Validating_and_Formatting_UK_Telephone_Numbers#Local_dialling_rules_for_UK_telephone_numbers

There&#039;s also a whole load of other useful information about the UK number plan on that site.</description>
		<content:encoded><![CDATA[<p>Anyone trying to make sense of their UK dialplan documentation should read the page at: <a href="http://www.aa-asterisk.org.uk/index.php/Errors_in_Cisco_UK_Numbering_Plan_Documents" rel="nofollow">http://www.aa-asterisk.org.uk/index.php/Errors_in_Cisco_UK_Numbering_Plan_Documents</a></p>
<p>If you want details of UK local dialling patterns within each area code, see the page at: <a href="http://www.aa-asterisk.org.uk/index.php/Regular_Expressions_for_Validating_and_Formatting_UK_Telephone_Numbers#Local_dialling_rules_for_UK_telephone_numbers" rel="nofollow">http://www.aa-asterisk.org.uk/index.php/Regular_Expressions_for_Validating_and_Formatting_UK_Telephone_Numbers#Local_dialling_rules_for_UK_telephone_numbers</a></p>
<p>There&#8217;s also a whole load of other useful information about the UK number plan on that site.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Cisco IP Phones 79XX with Asterisk &#124; Mehrdust Official Web Page</title>
		<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/#comment-338</link>
		<dc:creator>Cisco IP Phones 79XX with Asterisk &#124; Mehrdust Official Web Page</dc:creator>
		<pubDate>Fri, 25 Nov 2011 07:16:17 +0000</pubDate>
		<guid isPermaLink="false">http://minded.ca/?p=52#comment-338</guid>
		<description>[...] Configure Cisco IP Phones with Asterisk [...]</description>
		<content:encoded><![CDATA[<p>[...] Configure Cisco IP Phones with Asterisk [...]</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: kloklo</title>
		<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/#comment-332</link>
		<dc:creator>kloklo</dc:creator>
		<pubDate>Wed, 26 Oct 2011 14:43:15 +0000</pubDate>
		<guid isPermaLink="false">http://minded.ca/?p=52#comment-332</guid>
		<description>Hi,

I would like configure my CISCO phone with signed files. Do you know how I can do that?

Many thanks.</description>
		<content:encoded><![CDATA[<p>Hi,</p>
<p>I would like configure my CISCO phone with signed files. Do you know how I can do that?</p>
<p>Many thanks.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Tyler Winfield</title>
		<link>http://www.minded.ca/default/2009-12-16/configure-cisco-ip-phones-with-asterisk/#comment-148</link>
		<dc:creator>Tyler Winfield</dc:creator>
		<pubDate>Fri, 28 Jan 2011 16:22:11 +0000</pubDate>
		<guid isPermaLink="false">http://minded.ca/?p=52#comment-148</guid>
		<description>@Joe,

Getting a hold of the firmware is the big hurdle with this phones as downloading it requires a purchased Cisco license (which isn&#039;t the cheapest thing on the planet).  Spending some quality time with Google on a Sunday morning might just be what you need to get that firmware you seek.  A helpful note is that most firmware package filenames start with &quot;cmterm&quot; and contain the model number (&quot;7970&quot;) as well as the firmware type (&quot;sccp&quot;).  For asterisk, you&#039;ll want to look for .zip files; the .cop and other extensions are for use with CCM.

Once you are able to get a hold of the correct firmware; yes, the SEP.cnf.xml file will be required as described in this post.

Tyler</description>
		<content:encoded><![CDATA[<p>@Joe,</p>
<p>Getting a hold of the firmware is the big hurdle with this phones as downloading it requires a purchased Cisco license (which isn&#8217;t the cheapest thing on the planet).  Spending some quality time with Google on a Sunday morning might just be what you need to get that firmware you seek.  A helpful note is that most firmware package filenames start with &#8220;cmterm&#8221; and contain the model number (&#8220;7970&#8243;) as well as the firmware type (&#8220;sccp&#8221;).  For asterisk, you&#8217;ll want to look for .zip files; the .cop and other extensions are for use with CCM.</p>
<p>Once you are able to get a hold of the correct firmware; yes, the SEP.cnf.xml file will be required as described in this post.</p>
<p>Tyler</p>
]]></content:encoded>
	</item>
</channel>
</rss>

