When trying to connect from a sip-phone to the portal, <a href="http://sip.allstarlink.org">sip.allstarlink.org</a> using, <br><br>sip://<a href="http://1:allstar@sip.allstarlink.org:5060/0">1:allstar@sip.allstarlink.org:5060/0</a><br>
-or-<br>sip://<a href="http://1:allstar@sip.allstarlink.org:5060/">1:allstar@sip.allstarlink.org:5060/</a>&lt;node_number&gt;<br>-or-<br>sip://<a href="mailto:0@sip.allstarlink.org">0@sip.allstarlink.org</a><br><br>It connects, but there&#39;s no audio. No errors.  PCMU@8000 correctly negotiated.  I&#39;ve tried two SIP-phone clients and a FreeSWITCH user-agent with the same negative outcomes.  I&#39;ve tried variations on the connect stream with same negative results.  The BRIA soft-phone even handshakes GSM but still no audio.<br>
<br>Is there something I&#39;m missing?<br><br>Thanks,<br><br><br><br>