I guess the question here is, what is asterisk doing for you? I personally would prefer the sip trunks right on opensips.. Asterisk is a kinda funny bottleneck in your architecture unless it's acting as some sort of media server (or TDM gateway).<div>
<br></div><div>Some potential issues:</div><div>1. Do you have 2 way audio, some providers (gateways) will disco the call if there is one way audio for X seconds.</div><div>2. Do you see any reinvites happening? Some providers will re-invite calls after they are up and if the reinvite fails, it will tear down the call.</div>
<div>3. Where is the BYE coming from? Do you see any other signaling after the 200OK/ACKs you get? Do you see retransmissions of either the 200OK or ACK? If the signaling indicating the call was connected doesn't finish a proper ACK in both directions, the call will likely get hung up on.</div>
<div><br><br><div class="gmail_quote">On Tue, Oct 6, 2009 at 8:17 AM, Peter den Hartog <span dir="ltr"><<a href="mailto:peterdenhartog@gmail.com">peterdenhartog@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<br>
I'm trying to intergrate opensips with a allready running Asterisk server.<br>
The two servers are both on the same machine.<br>
<br>
I can recieve calls fine, Asterisk send them to my opensips installation,<br>
and the opensips forwards the phone call to the right user. I can call<br>
between the users on the network, with out any issue's so far so good.<br>
<br>
I have a sip trunk registered on Asterisk, and i use that for my in and<br>
outgoing calls.<br>
<br>
But when i make an outside call, the call ends after 17 seconds. Looking at<br>
the sip messages i see that i recieve a bye, then the call is gone.<br>
<br>
Am i doing something wrong, should the sip trunk be directly in opensips?<br>
and add that as a rewritehost? Or is this an Asterisk issue?<br>
<br>
My opensips is running on port 5090 (so are the phones) and my<br>
asterisk+outside trunk is on 5060.<br>
<font color="#888888">--<br>
View this message in context: <a href="http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3774964.html" target="_blank">http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3774964.html</a><br>
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.<br>
<br>
_______________________________________________<br>
Users mailing list<br>
<a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
<a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
</font></blockquote></div><br></div>