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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Greetings,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>You have stumbled on to
one of the most troublesome flag for <span class=SpellE>newbies</span>;
autocreatepeer.<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'><a
href="http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+autocreatepeer">http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+autocreatepeer</a><o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><span class=GramE><font size=2
color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'>in</span></font></span><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'> your sip.conf file add a
line in the [general] section “autocreatepeer=no”<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>Now people can only use
your Asterisk SIP connection if you create a peer entry for them in your
sip.conf file.<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>Your sip.conf file should
be located in “/etc/asterisk” directory.<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>“<span
class=SpellE><span class=GramE>cd</span></span> /etc/asterisk”<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>“<span class=GramE>vi</span>
sip.conf”<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Stojan Sljivic - GDS<br>
<b><span style='font-weight:bold'>Sent:</span></b> </span></font><st1:date
Month="6" Day="13" Year="2005"><font size=2 face=Tahoma><span style='font-size:
10.0pt;font-family:Tahoma'>Monday, June 13, 2005</span></font></st1:date><font
size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> </span></font><st1:time
Hour="2" Minute="18"><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma'>2:18 AM</span></font></st1:time><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'><br>
<b><span style='font-weight:bold'>To:</span></b> 'Asterisk Users Mailing List -
Non-Commercial Discussion'<br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: [Asterisk-Users] SIP
Authentication</span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'><o:p> </o:p></span></font></p>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>Hi,</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>Does
anyone know the solution to this issue?</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
</div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>Regards,<br>
Stojan Sljivic </span></font><o:p></o:p></p>
<blockquote style='border:none;border-left:solid maroon 1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>
<p class=MsoNormal style='mso-margin-top-alt:0in;margin-right:0in;margin-bottom:
12.0pt;margin-left:.5in'><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Stojan Sljivic - GDS<br>
<b><span style='font-weight:bold'>Sent:</span></b> </span></font><st1:date
Month="6" Day="10" Year="2005"><font size=2 face=Tahoma><span style='font-size:
10.0pt;font-family:Tahoma'>Friday, June 10, 2005</span></font></st1:date><font
size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> </span></font><st1:time
Hour="13" Minute="21"><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma'>13:21</span></font></st1:time><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'><br>
<b><span style='font-weight:bold'>To:</span></b> 'Asterisk Users Mailing List -
Non-Commercial Discussion'<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users] SIP
Authentication</span></font><o:p></o:p></p>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>Hi,</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>I use
SIP softphone that is not registered at Asterisk.</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>When I
dial some extension defined in the dial plan (<a href="mailto:100@asterisk-ip">100@asterisk-ip</a>) with
my SIP softphone, Asterisk will not ask me for username/password (will not
return response 407) as I expected.</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>The
response 407 - Authentication required will be returned if username defined in
the softphone's setting matches one of the SIP peers defined in sip.conf.</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>This
means that anyone can dial extension at my Asterisk and that is not good,
since that person could then dial over my ZAP line.</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>How
can I configure Asterisk to allow only peers defined in sip.conf to register
and dial?</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>Regards,</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=maroon
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:maroon'>Stojan
Sljivic</span></font><o:p></o:p></p>
</div>
</blockquote>
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