<html><head></head><body bgcolor="#FFFFFF"><div>Thanks for firewall pointers, but all appears in order in such regards. This isn't my site's first sip rodeo. We Have two working PBXs -- asterisk and FreeSWITCH -- up as well as an AS node and soft phones. </div>
<div><br></div><div>What has changed since first posting is that it now "just works" ,,, sometimes. It almost seems like a capacity thing on the allstarlink side. Now, I can dial, connect, no audio ... Hang up, redial, reconnect and it to works. Might be an early vs late negotiation skew.<br>
<br>Not sure why,,, if someone on the asl side has access to their sip logs, it might shed some light.</div><div><br></div><div>Thanks,<br>—————<div><div><br></div></div></div><div><br>On 2012-11-10, at 18:44, Scott Weis <<a href="mailto:kb2ear@kb2ear.net">kb2ear@kb2ear.net</a>> wrote:<br>
<br></div><div></div><blockquote type="cite"><div><meta http-equiv="Content-Type" content="text/html; charset=us-ascii"><meta name="Generator" content="Microsoft Word 14 (filtered medium)"><style><!--
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--></style><div class="WordSection1"><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Make sure you have the RTP ports forwarded in on your router.</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span></p><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:app_rpt-users-bounces@ohnosec.org">app_rpt-users-bounces@ohnosec.org</a> [mailto:<a href="mailto:app_rpt-users-bounces@ohnosec.org">app_rpt-users-bounces@ohnosec.org</a>] <b>On Behalf Of </b>Scott<br>
<b>Sent:</b> Saturday, November 10, 2012 5:45 PM<br><b>To:</b> <a href="mailto:App_rpt-users@ohnosec.org">App_rpt-users@ohnosec.org</a><br><b>Subject:</b> [App_rpt-users] <a href="http://sip.allstarlink.org">sip.allstarlink.org</a> sip session no-audio</span></p>
<p class="MsoNormal"> </p><p class="MsoNormal" style="margin-bottom:12.0pt">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><node_number><br>-or-<br>sip://<a href="mailto:0@sip.allstarlink.org">0@sip.allstarlink.org</a><br><br>It connects, but there's no audio. No errors. PCMU@8000 correctly negotiated. I've tried two SIP-phone clients and a FreeSWITCH user-agent with the same negative outcomes. I'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'm missing?<br><br>Thanks,<br><br><br></p></div></div></blockquote></body></html>