<div dir="ltr">Hi Alex<div><br></div><div>1.- Kamailio -> <a href="http://172.26.101.50:8002">172.26.101.50:8002</a> (Floating IP)<br><br>Asterisk -> <a href="http://172.26.101.10:5080">172.26.101.10:5080</a><div><br></div><div>2.- Transmitting (no NAT) to <a href="http://192.168.0.170:8002">192.168.0.170:8002</a>:</div>
<p class="p1"><span class="s1">ACK sip:110@IP_PUBLIC_IP:5066 SIP/2.0</span></p>
<p class="p1"><span class="s1">Via: SIP/2.0/UDP 172.26.101.10:5080;branch=z9hG4bK0a330ae6</span></p>
<p class="p1"><span class="s1">Route: <sip:192.168.0.170:8002;nat=no;ftag=as14d7523e;lr=on></span></p>
<p class="p1"><span class="s1">Max-Forwards: 70</span></p>
<p class="p1"><span class="s1">From: "asterisk" <<a href="http://sip:110@172.26.101.10:5080">sip:110@172.26.101.10:5080</a>>;tag=as14d7523e</span></p>
<p class="p1"><span class="s1">To: <<a href="http://sip:110@172.26.101.50:8002">sip:110@172.26.101.50:8002</a>>;tag=1749303708</span></p>
<p class="p1"><span class="s1">Contact: <<a href="http://sip:110@172.26.101.10:5080">sip:110@172.26.101.10:5080</a>></span></p>
<p class="p1"><span class="s1">Call-ID: <a href="http://7e723f1c64086e964df79e493350a2a4@172.26.101.10:5080">7e723f1c64086e964df79e493350a2a4@172.26.101.10:5080</a></span></p>
<p class="p1"><span class="s1">CSeq: 102 ACK</span></p>
<p class="p1"><span class="s1">User-Agent: ast01</span></p>
<p class="p1"><span class="s1">Content-Length: 0</span></p><p class="p1">With code i posted before i have now issue to answer calls from ast (generated by asterisk)</p><p class="p1"><br></p><p class="p1">200 From Phone arrives fine to Kamailio</p><p class="p1"><br></p><p class="p1"><span class="s1"><--- SIP read from UDP:<a href="http://172.26.101.50:8002">172.26.101.50:8002</a> ---></span></p><p class="p1"><span class="s1">SIP/2.0 200 OK</span></p><p class="p1"><span class="s1">Via: SIP/2.0/UDP 172.26.101.10:5080;rport=5080;branch=z9hG4bK125b3b98</span></p><p class="p1"><span class="s1">Record-Route: <sip:192.168.0.170:8002;nat=no;ftag=as14d7523e;lr=on></span></p><p class="p1"><span class="s1">From: "asterisk" <<a href="http://sip:110@172.26.101.10:5080">sip:110@172.26.101.10:5080</a>>;tag=as14d7523e</span></p><p class="p1"><span class="s1">To: <<a href="http://sip:110@172.26.101.50:8002">sip:110@172.26.101.50:8002</a>>;tag=1749303708</span></p><p class="p1"><span class="s1">Call-ID: <a href="http://7e723f1c64086e964df79e493350a2a4@172.26.101.10:5080">7e723f1c64086e964df79e493350a2a4@172.26.101.10:5080</a></span></p><p class="p1"><span class="s1">CSeq: 102 INVITE</span></p><p class="p1"><span class="s1">Contact: <sip:110@PUBLIC.IP:5066></span></p><p class="p1"><span class="s1">Supported: replaces, path, timer, eventlist</span></p><p class="p1"><span class="s1">User-Agent: Grandstream GXV3275 1.0.3.37</span></p><p class="p1"><span class="s1">Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE</span></p><p class="p1"><span class="s1">Content-Type: application/sdp</span></p><p class="p1"><span class="s1">Content-Length: 255</span></p><p class="p2"><span class="s1"></span><br></p><p class="p1"><span class="s1">v=0</span></p><p class="p1"><span class="s1">o=110 8003 8000 IN IP4 172.26.101.41</span></p><p class="p1"><span class="s1">s=SIP Call</span></p><p class="p1"><span class="s1">c=IN IP4 172.26.101.41</span></p><p class="p1"><span class="s1">t=0 0</span></p><p class="p1"><span class="s1">m=audio 8424 RTP/AVP 0 8 101</span></p><p class="p1"><span class="s1">a=sendrecv</span></p><p class="p1"><span class="s1">a=rtpmap:0 PCMU/8000</span></p><p class="p1"><span class="s1">a=ptime:20</span></p><p class="p1"><span class="s1">a=rtpmap:8 PCMA/8000</span></p><p class="p1"><span class="s1">a=rtpmap:101 telephone-event/8000</span></p><p class="p1"><span class="s1">a=fmtp:101 0-15</span></p><p class="p1"><span class="s1">a=sdp_proxied:yes</span></p><p class="p1">
</p><p class="p1"><span class="s1"><-------------></span></p><p class="p1"><span class="s1">I see 200 OK sent to Asterisk</span></p><p class="p1"><span class="s1"><br></span></p><p class="p1"><span class="s1">Asterisk sent to me ACK but Kamailio seems to do not send to Phone behind nat</span></p><p class="p1"><span class="s1"><br></span></p><p class="p1"><span class="s1">3.- <a href="http://192.168.0.170:8002">192.168.0.170:8002</a></span></p><p class="p1"><span class="s1"><br></span></p><p class="p1"><span class="s1">Thanks</span></p><div><br></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-07-29 9:18 GMT+02:00 Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com" target="_blank">abalashov@evaristesys.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Alberto,<br>
<br>
1. What are the literal (natively homed) IP addresses of Asterisk and Kamailio?<br>
<br>
2. What is the Request Line (first line) of the ACK request being sent from Asterisk, i.e.<br>
<br>
ACK sip:... SIP/2.0<br>
<br>
3. To what IP and port is the ACK being sent by Asterisk?<div class="HOEnZb"><div class="h5"><br>
<br>
-- <br>
Alex Balashov | Principal | Evariste Systems LLC<br>
303 Perimeter Center North, Suite 300<br>
Atlanta, GA 30346<br>
United States<br>
<br>
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)<br>
Web: <a href="http://www.evaristesys.com/" rel="noreferrer" target="_blank">http://www.evaristesys.com/</a>, <a href="http://www.csrpswitch.com/" rel="noreferrer" target="_blank">http://www.csrpswitch.com/</a><br>
<br>
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</div></div></blockquote></div><br></div>