<div dir="ltr"><div>Try this - <br></div><div><b><br>Polycom -</b><br></div><div><br></div><div>Add this to your config file. This will cause the phone to respond with a '400 bad request' unless the INVITE comes from the registrar. <br>
<br></div><div><span class="" style="background-color:rgb(255,255,204);color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.7273px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px">voIpProt</span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.7273px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);display:inline!important;float:none">.SIP.</span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.7273px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);display:inline!important;float:none">requestValidation.1.request="</span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.7273px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);display:inline!important;float:none">INVITE"<span class=""> </span></span><span class="" style="background-color:rgb(255,255,204);color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.7273px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px">voIpProt</span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.7273px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);display:inline!important;float:none">.SIP.</span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.7273px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);display:inline!important;float:none">requestValidation.1.method="</span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.7273px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);display:inline!important;float:none">source"<br>
<br></span><br><b>Adtran  (Only applies to the local SIP stack (voice users, trunks etc) )  </b><br><br>Add an ACL, which only lists your proxy/SBC IPs<br><br>ip access-list standard my-proxy<br> remark MY IP SPACE<br> permit  x.x.x.x  y.y.y.y log<br>
<br></div><div>
!Apply the ACL as a sip access-class<br></div>ip sip access-class my-proxy in<br><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Nov 12, 2013 at 4:37 PM, Jay Hennigan <span dir="ltr"><<a href="mailto:jay@west.net" target="_blank">jay@west.net</a>></span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">We operate a hosted VoIP platform using Broadworks and over 90% Polycom<br>
phones.  We have a number of customers at diverse locations recently<br>
reporting dead-air calls.  These typically come from a source ANI of 100<br>
or 7070 and have no audio.<br>
<br>
Most of the phones are behind Adtran TA900 series IADs with NAT and a<br>
SIP access list allowing only our own SBCs.<br>
<br>
The calls don't show up in our SBC logs, nor do they show in the MOS<br>
graphs on the Adtran voice quality monitoring.<br>
<br>
Any ideas or suggestions on what is causing this or how to stop it?<br>
<span><font color="#888888"><br>
--<br>
--<br>
Jay Hennigan - CCIE #7880 - Network Engineering - <a href="mailto:jay@impulse.net" target="_blank">jay@impulse.net</a><br>
Impulse Internet Service  -  <a href="http://www.impulse.net/" target="_blank">http://www.impulse.net/</a><br>
Your local telephone and internet company - <a href="tel:805%20884-6323" value="+18058846323" target="_blank">805 884-6323</a> - WB6RDV<br>
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</font></span></blockquote></div><br></div></div>