Merged revisions 171837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -49,6 +49,25 @@
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;
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;
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; module reload chan_sip.so Reload configuration file
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; module reload chan_sip.so Reload configuration file
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;
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;
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;------- Naming devices ------------------------------------------------------
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;
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; When naming devices, make sure you understand how Asterisk matches calls
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; that come in.
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; 1. Asterisk checks the SIP From: address username and matches against
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; names of devices with type=user
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; The name is the text between square brackets [name]
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; 2. Asterisk checks the IP address (and port number) that the INVITE
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; was sent from and matches against any devices with type=peer
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;
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; Don't mix extensions with the names of the devices. Devices need a unique
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; name. The device name is *not* used as phone numbers. Phone numbers are
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; anything you declare as an extension in the dialplan (extensions.conf).
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;
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; Note: The parameter "username" is not the username and in most cases is
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; not needed at all. Check below. In later releases, it's renamed
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; to "defaultuser" which is a better name, since it is used in
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; combination with the "defaultip" setting.
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;-----------------------------------------------------------------------------
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; ** Deprecated configuration options **
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; ** Deprecated configuration options **
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; The "call-limit" configuation option is deprecated. It still works in
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; The "call-limit" configuation option is deprecated. It still works in
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