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	<title>VoIP Cookbook: Defining SIP Channel in sip.conf - Revision history</title>
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	<updated>2026-04-20T16:08:46Z</updated>
	<subtitle>Revision history for this page on the wiki</subtitle>
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		<id>https://lms.onnocenter.or.id/wiki/index.php?title=VoIP_Cookbook:_Defining_SIP_Channel_in_sip.conf&amp;diff=17275&amp;oldid=prev</id>
		<title>Onnowpurbo: New page: Every SIP client and server is identified with a text block such as the following:    [xxx]   type=yyy   parameter1=value   parameter2=value   Where “xxx” is the name associated with t...</title>
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		<updated>2010-03-09T02:59:46Z</updated>

		<summary type="html">&lt;p&gt;New page: Every SIP client and server is identified with a text block such as the following:    [xxx]   type=yyy   parameter1=value   parameter2=value   Where “xxx” is the name associated with t...&lt;/p&gt;
&lt;p&gt;&lt;b&gt;New page&lt;/b&gt;&lt;/p&gt;&lt;div&gt;Every SIP client and server is identified with a text block such as the following: &lt;br /&gt;
&lt;br /&gt;
 [xxx] &lt;br /&gt;
 type=yyy &lt;br /&gt;
 parameter1=value &lt;br /&gt;
 parameter2=value &lt;br /&gt;
&lt;br /&gt;
Where “xxx” is the name associated with the SIP client, or any arbitrary name used by other configuration file to refer to a SIP device. Typically, a SIP phone with extension number 123 will have an entry that begins with [123]. Please note that you still have to activate extension 123 in extensions.conf file so that people can call to extension 123. The parameter Type must contain &amp;quot;user&amp;quot;, &amp;quot;peer&amp;quot; or &amp;quot;friend&amp;quot;. &lt;br /&gt;
&lt;br /&gt;
Asterisk will match the incoming call with the name of the device and type=user of the SIP protocol conversation in column From: user name (without acknowledging the SIP domain). Another way also used is to match the SIP request and [xxx] in sip.conf file, through both the IP address of the requester and the peer information of part of [xxx] in sip.conf file. If host=dynamic, it&amp;#039;s not possible to perform matching until the SIP client is registered. &lt;br /&gt;
&lt;br /&gt;
==See Also==&lt;br /&gt;
&lt;br /&gt;
* [[VoIP Cookbook: Building your own Telecommunication Infrastructure]]&lt;br /&gt;
* [[VoIP Cookbook: Asterisk for Incoming and Outgoing calls]]&lt;/div&gt;</summary>
		<author><name>Onnowpurbo</name></author>
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