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<DIV><SPAN class=801181907-12112009><FONT size=2 face=Arial>At Tilghman's 
request.</FONT></SPAN></DIV>
<DIV><SPAN class=801181907-12112009><FONT size=2 
face=Arial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=801181907-12112009><FONT size=2 face=Arial>We need to agree to 
change the sip.conf default from allowguest=yes to 
allowguest=no</FONT></SPAN></DIV>
<DIV><SPAN class=801181907-12112009><FONT size=2 face=Arial>and extensions.conf 
to have a warning in the [default] section that sip.conf may have allowguest=yes 
or nothing which will&nbsp;default of yes.</FONT></SPAN></DIV>
<DIV><SPAN class=801181907-12112009></SPAN><SPAN 
class=801181907-12112009></SPAN><SPAN class=801181907-12112009><FONT size=2 
face=Arial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=801181907-12112009><FONT size=2 
face=Arial>Reference&nbsp;mantis bugs;</FONT></SPAN></DIV>
<DIV><SPAN class=801181907-12112009><A 
href="https://issues.asterisk.org/view.php?id=15101"><FONT size=2 
face=Arial>https://issues.asterisk.org/view.php?id=15101</FONT></A><FONT size=2 
face=Arial>&nbsp;SIP allowguest defaults to yes with 'make samples' 
</FONT></SPAN></DIV>
<DIV><SPAN class=801181907-12112009><A 
href="https://issues.asterisk.org/view.php?id=16226"><FONT size=2 
face=Arial>https://issues.asterisk.org/view.php?id=16226</FONT></A><FONT size=2 
face=Arial>&nbsp;1.4.26.3 security issue - Chinese IPs somehow are making calls 
without authentication </FONT></SPAN></DIV>
<DIV><SPAN class=801181907-12112009><FONT size=2 
face=Arial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=801181907-12112009><FONT size=2 face=Arial>There are many 
installations out there where&nbsp;newbies are playing in the [default] context 
in their dialplan, getting things working, then opening port 5060 in their 
firewall without understanding what they've just done.</FONT></SPAN></DIV>
<DIV><SPAN class=801181907-12112009><FONT size=2 
face=Arial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=801181907-12112009><FONT size=2 face=Arial>Initially 
I&nbsp;thought it was&nbsp;great that we allow any&nbsp;SIP phone to connect to 
asterisk, with no configuration required at the astrisk end, how wrong I 
was.&nbsp;</FONT></SPAN></DIV>
<DIV><SPAN class=801181907-12112009><FONT size=2 
face=Arial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=801181907-12112009><FONT size=2 face=Arial>Alec 
Davis</FONT></SPAN></DIV></BODY></HTML>