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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'>You may need to look to see if you are
using peer or friend in the sip config for this phone. We needed to change ours
to friend to make it work for us with still using secret.<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>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
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> Sunday, June 12, 2005 11:18
PM<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><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><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>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><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>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><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;
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<p class=MsoNormal style='margin-bottom:12.0pt'><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> Friday, June 10, 2005 13:21<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>
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<p class=MsoNormal><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>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><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>
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<p class=MsoNormal><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>
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<p class=MsoNormal><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>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><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>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><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>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><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>
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<p class=MsoNormal><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>
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