<div dir="ltr">I am trying to verify that SIP TLS calls work through FreeSwitch running on Windows 7.<div><br></div><div>But the outgoing leg of the call does not get established by FreeSwitch.</div><div><br></div><div>Setup:</div>
<div><br></div><div>1 PC running Windows 7 and FreeSwitch version 1.4.7 </div><div><br></div><div>FSClient 1.2.3.5 is installed on two PCs and are configured & registered via TLS to the FreeSwitch server.</div><div><br>
</div><div>When I make a call from one FSClient (1003) to the other (1005), I see that the call request gets to the FreeSwitch server:</div><div><br></div><div><div>recv 1600 bytes from tls/[10.47.26.44]:14183 at 19:56:53.257366:</div>
<div> ------------------------------------------------------------------------</div><div> INVITE <a href="mailto:sip%3A1005@sszetofs1a.design.mitel.com">sip:1005@sszetofs1a.design.mitel.com</a>;transport=tls SIP/2.0</div>
<div><div> From: "x1003 tls" <<a href="mailto:sip%3A1003@sszetofs1a.design.mitel.com">sip:1003@sszetofs1a.design.mitel.com</a>>;tag=cKD7B99Z9124m</div><div> To: <<a href="mailto:sip%3A1005@sszetofs1a.design.mitel.com">sip:1005@sszetofs1a.design.mitel.com</a>></div>
</div><div><br></div><div><br></div><div><div> ------------------------------------------------------------------------</div><div>send 353 bytes to tls/[10.47.26.44]:14183 at 19:56:53.257366:</div><div> ------------------------------------------------------------------------</div>
<div> SIP/2.0 100 Trying</div><div> Via: SIP/2.0/TLS 10.47.26.44:12347;branch=z9hG4bKpKHKUN4jFUerN;rport=14183</div><div> From: "x1003 tls" <<a href="mailto:sip%3A1003@sszetofs1a.design.mitel.com">sip:1003@sszetofs1a.design.mitel.com</a>>;tag=cKD7B99Z9124m</div>
<div> To: <<a href="mailto:sip%3A1005@sszetofs1a.design.mitel.com">sip:1005@sszetofs1a.design.mitel.com</a>></div></div><div><br></div><div><br></div><div><br></div><div>The call gets challenged and authenticated:</div>
<div><br></div><div><div> SIP/2.0 407 Proxy Authentication Required</div><div> Via: SIP/2.0/TLS 10.47.26.44:12347;branch=z9hG4bKpKHKUN4jFUerN;rport=14183</div><div> From: "x1003 tls" <<a href="mailto:sip%3A1003@sszetofs1a.design.mitel.com">sip:1003@sszetofs1a.design.mitel.com</a>>;tag=cKD7B99Z9124m</div>
<div> To: <<a href="mailto:sip%3A1005@sszetofs1a.design.mitel.com">sip:1005@sszetofs1a.design.mitel.com</a>>;tag=6B6B175KHK8mN</div></div><div><br></div><div><br></div><div><div> ACK <a href="mailto:sip%3A1005@sszetofs1a.design.mitel.com">sip:1005@sszetofs1a.design.mitel.com</a>;transport=tls SIP/2.0</div>
<div> Via: SIP/2.0/TLS 10.47.26.44:12347;branch=z9hG4bKpKHKUN4jFUerN</div><div> Max-Forwards: 70</div><div> From: "x1003 tls" <<a href="mailto:sip%3A1003@sszetofs1a.design.mitel.com">sip:1003@sszetofs1a.design.mitel.com</a>>;tag=cKD7B99Z9124m</div>
<div> To: <<a href="mailto:sip%3A1005@sszetofs1a.design.mitel.com">sip:1005@sszetofs1a.design.mitel.com</a>>;tag=6B6B175KHK8mN</div></div><div><br></div><div><br></div><div>I then see INVITE/100 Trying being exchanged between the caller and FreeSwitch:</div>
<div><br></div><div> INVITE <a href="mailto:sip%3A1005@sszetofs1a.design.mitel.com">sip:1005@sszetofs1a.design.mitel.com</a>;transport=tls SIP/2.0<br></div><div><div> Via: SIP/2.0/TLS 10.47.26.44:12347;branch=z9hG4bKQvacXgNpc44aH</div>
<div> Max-Forwards: 70</div><div> From: "x1003 tls" <<a href="mailto:sip%3A1003@sszetofs1a.design.mitel.com">sip:1003@sszetofs1a.design.mitel.com</a>>;tag=cKD7B99Z9124m</div><div> To: <<a href="mailto:sip%3A1005@sszetofs1a.design.mitel.com">sip:1005@sszetofs1a.design.mitel.com</a>></div>
</div><div><br></div><div><br></div><div><div> SIP/2.0 100 Trying</div><div> Via: SIP/2.0/TLS 10.47.26.44:12347;branch=z9hG4bKQvacXgNpc44aH;rport=14183</div><div> From: "x1003 tls" <<a href="mailto:sip%3A1003@sszetofs1a.design.mitel.com">sip:1003@sszetofs1a.design.mitel.com</a>>;tag=cKD7B99Z9124m</div>
<div> To: <<a href="mailto:sip%3A1005@sszetofs1a.design.mitel.com">sip:1005@sszetofs1a.design.mitel.com</a>></div></div><div><br></div><div><br></div><div><br></div><div>I see a new channel being created to try and reach the terminating party (1005):</div>
<div><br></div><div><div>2014-08-20 15:56:53.387379 [INFO] mod_dialplan_xml.c:558 Processing x1003 tls <1003>->1005 in context default</div><div>2014-08-20 15:56:53.467387 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/<a href="http://sip:gw+3@10.47.26.34:12347">sip:gw+3@10.47.26.34:12347</a> [d416a13a-fb99-4eb4-8645-24104ad5ed97]</div>
</div><div><br></div><div><br></div><div>But that's it ... no SIP messages are sent to 1005</div><div><br></div><div>Any thoughts? Are there any parameters I can set within FreeSwitch to make the outbound call leg work?</div>
<div><br></div><div><br></div>-- <br><div dir="ltr"><table border="0" cellspacing="0" cellpadding="0" style="border-collapse:collapse"><tbody><tr><td width="216" valign="top" style="width:161.85pt;padding:0in"><p><b><span style="font-size:8pt;font-family:Verdana">Regards,</span></b></p>
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