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On 9/11/15 12:59 PM, Jerry Geis wrote:<br>
<blockquote
cite="mid:CABr8-B6ryVkmfPasCtAaheEUmt8nVOHa6MNd8WNvECEM0n3qWw@mail.gmail.com"
type="cite">
<div dir="ltr">I have a setup where I have polycom phones in an
office, behind firewall,
<div>going out to a server located elsewhere. I have set </div>
<div>nat=force_rport,comedia for my phones.</div>
<div><br>
</div>
<div>so if I call OUT to my cell I get audio both ways and the
call is fine.</div>
<div><br>
</div>
<div>My issue is if I call phone to phone in the office the
phone doesnt</div>
<div>even ring. The CLI shows I'm calling the correct extension
like SIP/524.</div>
<div><br>
</div>
<div>I am using asterisk 11.19.0</div>
<div><br>
</div>
<div>Is there another setting to correctly to this type of
calling?</div>
<div><br>
</div>
<div>Thanks,</div>
<div><br>
</div>
<div>Jerry</div>
</div>
<br>
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</blockquote>
There are usually two issues that can cause this behaviour. One
is that you do not have a correct "localnet" definition and the
other is that you have directmedia=yes on your sip.conf.<br>
<br>
<pre class="moz-signature" cols="72">--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161</pre>
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