<div dir="ltr">Vlad,<div style>Thanks for responding. Unfortunately, I made the suggested change and still have the same results. Here is some more information. I have added a new sip trace and opensips log.</div><div style>
<br></div><div style>Thanks,</div><div style>Brad</div><div style><br></div><div style>The call is origianted from (7278516359)</div><div style>The caller dials 8665551212</div><div style>The call is sent to 63.246.156.XX (opensips 1.8.1)</div>
<div style>There the call is forwarded to 192.168.1.21 from 192.168.1.22 (opensips second interface)</div><div style><br></div><div style>Asterisk answers, dials 7275551212 and then Bridges the two calls.</div><div style>
<br></div><div style>7275551212 hangs up first and the problem arises.</div><div style><br></div><div style>If 7278516359 hangs up first, the call terminates correctly.</div><div style><br></div><div style>One thing I did notice, when 7278516359 hangs up the sip trace shows the path:</div>
<div style>ISP ---> opensips (public IP) --> Astersik (private IP)</div><div style><br></div><div style>When 727551212 hangs up first, the path is as follows.</div><div style>ISP ---> opensips (public IP) --> Opensips (public IP) --> 404 not here<br>
</div><div style><br></div><div style>opensips log</div><div style><a href="https://gist.github.com/anonymous/5030013">https://gist.github.com/anonymous/5030013</a><br></div><div style><br></div><div style>sip trace</div>
<div style><a href="https://gist.github.com/5030094">https://gist.github.com/5030094</a><br></div><div style><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Feb 25, 2013 at 8:01 AM, Vlad Paiu <span dir="ltr"><<a href="mailto:vladpaiu@opensips.org" target="_blank">vladpaiu@opensips.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><u></u>
<div bgcolor="#ffffff" text="#000000">
Hello,<br>
<br>
Seems the incoming BYE does not have any Route headers, and the
loose_route() function returns false.<br>
<br>
Since you have dialog support in your script, try<br>
<pre><div> if (has_totag()) {</div><div> # sequential request withing a dialog should</div><div> # take the path determined by record-routing</div><div> if (loose_route() || match_dialog()) {
</div></pre>
This way you will force matching of dialog sequential requests that
have no Route headers.<br>
<br>
Best Regards,<br>
<pre cols="72">Vlad Paiu
OpenSIPS Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a> </pre><div><div class="h5">
<br>
On 02/24/2013 02:57 AM, brad smith wrote:
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr"><span style="color:rgb(85,85,85);font-family:arial,helvetica,clean,sans-serif;font-size:13px;line-height:18px;white-space:pre-wrap">Hello,<br>
I am currently running opensips 1.8.1 no tls. It is
multi-homed with a public and private address. </span>
<div><span style="color:rgb(85,85,85);font-family:arial,helvetica,clean,sans-serif;font-size:13px;line-height:18px;white-space:pre-wrap">I have a asterisk
1.8.19 in the lan that is connected to opensips via lan
address. </span></div>
<div><span style="color:rgb(85,85,85);font-family:arial,helvetica,clean,sans-serif;font-size:13px;line-height:18px;white-space:pre-wrap"><br>
</span></div>
<div><font color="#555555" face="arial, helvetica,
clean, sans-serif"><span style="line-height:18px;white-space:pre-wrap">*issue* </span></font></div>
<div><span style="color:rgb(85,85,85);font-family:arial,helvetica,clean,sans-serif;font-size:13px;line-height:18px;white-space:pre-wrap">A caller calls in
and then I place an outbound call and finally bridge the two
calls. </span></div>
<div><span style="color:rgb(85,85,85);font-family:arial,helvetica,clean,sans-serif;font-size:13px;line-height:18px;white-space:pre-wrap">This works as
expected, except when the outbound caller hangs up first the
BYE never gets back to Asterisk. </span></div>
<div><span style="color:rgb(85,85,85);font-family:arial,helvetica,clean,sans-serif;font-size:13px;line-height:18px;white-space:pre-wrap">I can see the BYE
reach OpenSips but a '404 not here' is returned to the ISP.
<br>
<br>
sip trace </span><a href="https://gist.github.com/5009662" style="color:rgb(0,102,153);text-decoration:none;margin:0px;padding:0px;font-size:13px;vertical-align:baseline;outline:medium none;font-family:arial,helvetica,clean,sans-serif;line-height:18px;white-space:pre-wrap" target="_blank">https://gist.github.com/5009662</a><span style="color:rgb(85,85,85);font-family:arial,helvetica,clean,sans-serif;font-size:13px;line-height:18px;white-space:pre-wrap"> <br>
opensips.cfg </span><a href="https://gist.github.com/5009704" style="color:rgb(0,102,153);text-decoration:none;margin:0px;padding:0px;font-size:13px;vertical-align:baseline;outline:medium none;font-family:arial,helvetica,clean,sans-serif;line-height:18px;white-space:pre-wrap" target="_blank">https://gist.github.com/5009704</a><span style="color:rgb(85,85,85);font-family:arial,helvetica,clean,sans-serif;font-size:13px;line-height:18px;white-space:pre-wrap"><br>
<br>
<br>
thanks for your time.</span><br>
</div>
</div>
</div></div><pre><fieldset></fieldset>
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</blockquote>
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