<p dir="ltr">It seems purely a NAT handling issue. While registering callee phone the NAT test should store routable/reachable contact and use it to send call. However, it is directly sending call to private IP and hence timeout occurs. </p>
<p dir="ltr">Do print some log lines while doing uac_nat_test while registering. Also use opensipsctl command line tool to see details of online users..AFAIR its "opensipsctl ul show" <br>
See what flags are set for the callee. <br>
</p>
<div class="gmail_quote">On Aug 6, 2015 12:14 AM, "Nabeel" <<a href="mailto:nabeelshikder@gmail.com">nabeelshikder@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div><br></div><div>Yes, the destination IP is <span style="font-size:12.8000001907349px"> </span><span style="color:rgb(0,0,0);font-family:Consolas,Menlo,Monaco,'Lucida Console','Liberation Mono','DejaVu Sans Mono','Bitstream Vera Sans Mono',monospace,serif;font-size:12px;line-height:21px"><a href="http://192.168.0.19:60912/" target="_blank">192.168.0.19:60912</a> and both phones are registered to OpenSIPS. In this case, the callee is connected to Wi-Fi (hence 192.xx IP address) and the caller is connected to a mobile network.</span></div><div><span style="color:rgb(0,0,0);font-family:Consolas,Menlo,Monaco,'Lucida Console','Liberation Mono','DejaVu Sans Mono','Bitstream Vera Sans Mono',monospace,serif;font-size:12px;line-height:21px"><br></span></div><div><font color="#000000" face="Consolas, Menlo, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, monospace, serif"><span style="font-size:12px;line-height:21px">The opensips.cfg I am using was generated from 'make menuconfig', except with the addition of "alias=<a href="http://domain.com" target="_blank">domain.com</a>". I have attached my config file at this link:</span></font></div><div><font color="#000000" face="Consolas, Menlo, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, monospace, serif"><span style="font-size:12px;line-height:21px"><br></span></font></div><div><font color="#000000" face="Consolas, Menlo, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, monospace, serif"><span style="font-size:12px;line-height:21px"><a href="http://pastebin.com/0QRyC938" target="_blank">http://pastebin.com/0QRyC938</a></span><br></font></div><div><font color="#000000" face="Consolas, Menlo, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, monospace, serif"><span style="font-size:12px;line-height:21px"><br></span></font></div><div><font color="#000000" face="Consolas, Menlo, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, monospace, serif"><span style="font-size:12px;line-height:21px"><br></span></font></div></div><div class="gmail_extra"><br><div class="gmail_quote">On 6 August 2015 at 05:00, SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi Nabeel,<div>Quick question; what is this destination ip? <span style="color:rgb(0,0,0);font-family:Consolas,Menlo,Monaco,'Lucida Console','Liberation Mono','DejaVu Sans Mono','Bitstream Vera Sans Mono',monospace,serif;font-size:12px;line-height:21px"><a href="http://192.168.0.19:60912" target="_blank">192.168.0.19:60912</a> ?</span><span style="color:rgb(0,0,0);font-family:Consolas,Menlo,Monaco,'Lucida Console','Liberation Mono','DejaVu Sans Mono','Bitstream Vera Sans Mono',monospace,serif;font-size:12px;line-height:21px"> - </span>Destination User Agent Registered on OpenSIPS?</div><div>Can you share the opensips.cfg code snippet for this call ?</div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Wed, Aug 5, 2015 at 11:55 PM, Nabeel <span dir="ltr"><<a href="mailto:nabeelshikder@gmail.com" target="_blank">nabeelshikder@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><div>Hi,</div><div><br></div><div>I am using the residential script generated by 'make menuconfig', with UDP and NAT support enabled. I added "alias=<a href="http://domain.com" target="_blank">domain.com</a>" to the config because otherwise the UA did not register with my domain (<a href="mailto:username@domain.com" target="_blank">username@domain.com</a>). When I attempt to make a call, I see '408 Request Timeout' in the sip trace and the call does not connect. Please check the log/trace below and advise how to fix this.</div><div><br></div><div>SIP trace:</div><div><br></div><a href="http://pastebin.com/u5h9qGNr" target="_blank">http://pastebin.com/u5h9qGNr</a><br><div><br></div><div>OpenSIPS log: </div><div><br></div><div><a href="http://pastebin.com/B8PUCKh0" target="_blank">http://pastebin.com/B8PUCKh0</a><br></div></div>
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