<?xml version="1.0" encoding="UTF-8"?><rss xmlns:dc="http://purl.org/dc/elements/1.1/" xmlns:content="http://purl.org/rss/1.0/modules/content/" xmlns:atom="http://www.w3.org/2005/Atom" version="2.0"><channel><title><![CDATA[SIP.conf/fop2 users, how do you tie them up?]]></title><description><![CDATA[<p dir="auto">I'm sure this is staring me in the face, but I don't quite understand the relationship between the user list in fop2.cfg and Sip.conf user list/sip registrations , how do you tie them together?(they are presumably the same thing(?))</p>
]]></description><link>http://localhost:4567/topic/817/sip.conf-fop2-users-how-do-you-tie-them-up</link><generator>RSS for Node</generator><lastBuildDate>Tue, 27 Oct 2026 08:04:58 GMT</lastBuildDate><atom:link href="http://localhost:4567/topic/817.rss" rel="self" type="application/rss+xml"/><pubDate>Mon, 27 Jun 2011 18:21:12 GMT</pubDate><ttl>60</ttl><item><title><![CDATA[Reply to SIP.conf/fop2 users, how do you tie them up? on Wed, 29 Jun 2011 14:53:46 GMT]]></title><description><![CDATA[<p dir="auto">Thanks, that was it.</p>
]]></description><link>http://localhost:4567/post/3639</link><guid isPermaLink="true">http://localhost:4567/post/3639</guid><dc:creator><![CDATA[silas2]]></dc:creator><pubDate>Wed, 29 Jun 2011 14:53:46 GMT</pubDate></item><item><title><![CDATA[Reply to SIP.conf/fop2 users, how do you tie them up? on Mon, 27 Jun 2011 21:32:54 GMT]]></title><description><![CDATA[<p dir="auto">I do not understand what is your doubt:</p>
<p dir="auto"><a href="http://www.fop2.com/documentation-configuration-buttons.php" rel="nofollow ugc">http://www.fop2.com/documentation-confi ... uttons.php</a></p>
<p dir="auto">In Asterisk you have device names, fop2 monitors those device names.. The device name depends on the channel driver. That is outside the scope of fop2 itself, is an asterisk thingy... so, sip channels in asterisk are named:</p>
<p dir="auto">SIP/xxxxx</p>
<p dir="auto">In sip.conf you can define a peer like</p>
<p dir="auto">[100]</p>
<p dir="auto">And the device name when in use in asterisk will be</p>
<p dir="auto">SIP/100</p>
<p dir="auto">In fop2.cfg you  must use the device/channel name. Example:</p>
<p dir="auto">SIP/100<br />
IAX2/mary<br />
DHADI/2</p>
<p dir="auto">The documentation is in the link I posted above, or you can see it also in the buttons.cfg.sample included with fop2</p>
]]></description><link>http://localhost:4567/post/3638</link><guid isPermaLink="true">http://localhost:4567/post/3638</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Mon, 27 Jun 2011 21:32:54 GMT</pubDate></item><item><title><![CDATA[Reply to SIP.conf/fop2 users, how do you tie them up? on Mon, 27 Jun 2011 21:10:44 GMT]]></title><description><![CDATA[<p dir="auto">Is it that the buttons bind to channels, which take the form SIP/username (=sip.conf username??)<br />
(link to relevant documentation would be welcome if I'm overlooking the obvious)</p>
]]></description><link>http://localhost:4567/post/3637</link><guid isPermaLink="true">http://localhost:4567/post/3637</guid><dc:creator><![CDATA[silas2]]></dc:creator><pubDate>Mon, 27 Jun 2011 21:10:44 GMT</pubDate></item><item><title><![CDATA[Reply to SIP.conf/fop2 users, how do you tie them up? on Mon, 27 Jun 2011 19:23:03 GMT]]></title><description><![CDATA[<p dir="auto">I do not understand your question. You define buttons in fop2 as in the example above, it is not a fop2 user but a fop2 button.</p>
<p dir="auto">The button definition requires the device to monitor between brackets, that is the "bond" that you are looking for with your asterisk sip.conf file.. but it is not only with that, as you can have other button types, like conferences, queues, etc..</p>
]]></description><link>http://localhost:4567/post/3636</link><guid isPermaLink="true">http://localhost:4567/post/3636</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Mon, 27 Jun 2011 19:23:03 GMT</pubDate></item><item><title><![CDATA[Reply to SIP.conf/fop2 users, how do you tie them up? on Mon, 27 Jun 2011 18:59:16 GMT]]></title><description><![CDATA[<p dir="auto">If know I must be exposing a fundamental misunderstanding, but if asterisk has all these sip calls getting 'bridged' between the registered sip phones and the outside world, doesn't fop2 somehow 'sit' on top of that to flash the lines which are active, etc, etc?</p>
]]></description><link>http://localhost:4567/post/3635</link><guid isPermaLink="true">http://localhost:4567/post/3635</guid><dc:creator><![CDATA[silas2]]></dc:creator><pubDate>Mon, 27 Jun 2011 18:59:16 GMT</pubDate></item><item><title><![CDATA[Reply to SIP.conf/fop2 users, how do you tie them up? on Mon, 27 Jun 2011 18:39:23 GMT]]></title><description><![CDATA[<p dir="auto">Hi,</p>
<p dir="auto">The user list in fop2.cfg is for defining extensions that you can use to log into fop2 itself.</p>
<p dir="auto">The user must be the extension number if you want to be able to perform actions in fop2. You can use anything else too, strings, invented numbers, but in those cases you won't be able to perform actions.</p>
<p dir="auto">There is no direct relation to sip.conf, that is up to your own asterisk configuration.  The relation for fop2 is between the user number and the extension setting in a button configuration also in fop2.cfg</p>
<p dir="auto">Example</p>
<p dir="auto">fop2.cfg</p>
<pre><code>user=100:1234:all
buttonfile=buttons.cfg
</code></pre>
<p dir="auto">buttons.cfg</p>
<pre><code>[SIP/john]
type=extension
extension=100
context=from-internal
</code></pre>
<p dir="auto">So, as you can see, there is a relation between the user and the button. But no relation at all with your sip.conf</p>
]]></description><link>http://localhost:4567/post/3634</link><guid isPermaLink="true">http://localhost:4567/post/3634</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Mon, 27 Jun 2011 18:39:23 GMT</pubDate></item></channel></rss>