<div dir="ltr">Bogdan,<div style>Thanks for responding. </div><div style>I am using vitelity for my upstream; I will send them a ticket. If they fail to act, do you have any suggestions...switch carriers? any config change?</div>
<div style><br></div><div style>Thanks again,</div><div style>Brad</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Feb 26, 2013 at 7:58 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@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">
<tt>Hi Brad,<br>
<br>
Thinks are a bit more complicated, it seems....<br>
<br>
In the INVITE your opensips sends to 64.....93 IP, you have the
Contact with 192.168.1.21 (priv IP of asterisk).<br>
<br>
When you receive the BYE from </tt><tt>64.....93 IP, the Route
hdrs are ok (the 2 hdrs added by opensips to reflect the interface
exchange), but the RURI is wrong - it must be the contact from the
INVITE you sent, but it seems to be the IP of your opensips - this
makes opensips to do act as strict router and not like a loose
router....and routing gets broken.<br>
<br>
So, the </tt><tt>64.....93 party or some other behind it, screw
up the Contact in the your INVITE and this alters the in-dialog
requests - you should check with the upstream guys.<br>
<br>
Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre><div><div class="h5">
<br>
On 02/25/2013 04:36 PM, brad smith wrote:
<blockquote type="cite">
<div dir="ltr">I just tested an outbound call (Asterisk originate)
without bridging and get the same '404 not here' if that helps.
<div><br>
</div>
<div>Thanks again,</div>
<div>Brad</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:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<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> <br>
On 02/24/2013 02:57 AM, brad smith wrote: </div>
</div>
<blockquote type="cite">
<div>
<div>
<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>
<br>
I am currently running opensips 1.8.1 no tls. It
is<br>
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<br>
1.8.19 in the lan that is connected to
opensips via lan<br>
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>
<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<br>
and then I place an outbound call and finally
bridge the two<br>
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<br>
expected, except when the outbound caller
hangs up first the<br>
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<br>
reach OpenSips but a '404 not here' is
returned to the ISP.<br>
<br>
<br>
<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>
<br>
<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>
<br>
<br>
<br>
<br>
thanks for your time.</span><br>
</div>
</div>
</div>
</div>
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