Hi,<br><br><div class="gmail_quote">On Sat, Oct 8, 2011 at 2:09 AM, caio <span dir="ltr">&lt;<a href="mailto:elcaio@gmail.com">elcaio@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hello,<br><br>I have the following issue.<br>A calls B, talk for a while, then B sends a BYE, but A never receives it. The BYE is transmited from the proxy listening interface:port to itself, but not to the asterisk port.<br>

<br>     U 2011/10/08 00:29:29.207335 Kamailio_IP:5060 -&gt; Kamailio_IP:5060<br>     BYE sip:Kamailio_IP;lr=on SIP/2.0.<br><br>The scenario is:<br><br>  A (pstn subs) --&gt; Asterisk:5080 --&gt; Kamailio:5060 --&gt; B (registered sip ua)<br>

<br>Asterisk and Kamailio are in the same box, listening both at the same IP (eth iface) address, but one at port 5080, and the proxy at 5060 (udp).<br></blockquote><div><br><br>Fixed changing the listen ip address of asterisk service. I set sip-router at eth0/&lt;ip_addr1&gt;:5060 and asterisk service at eth0:0/&lt;ip_addr2&gt;:5080.<br>
<br>Do you know about any issue keeping sip-router and asterisk at same interface but different port?<br>Thank you<br><br> </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
The configuration is based on kamailio.cfg file provided by the v3.1.5 (with a few modules added, like dialplan and cr..etc).<br>
<br>If A is who send the BYE, everything goes well.<br><br>I attach the full sip capture.<br>Thanks for you time.<br><br>Claudio<br>
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