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Merged revisions 175952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines Merged revisions 175921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -64,7 +64,7 @@ Autoprovisioning :
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- This feature must only be used on a trusted network. It's very insecure : all unistim phones
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will be able to use your asterisk pbx.
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- You must add an entry called [template]. Each new phones will be based on this profile.
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- You must set a least line=>. This value will be incremented when a new phone is registred.
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- You must set a least line=>. This value will be incremented when a new phone is registered.
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device= must not be specified. By default, the phone will asks for a number. It will be added into
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the dialplan. Add extension=line for using the generated line number instead.
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Example :
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