Usually, calls terminating at 30 seconds is a sure sign that you need to add an Answer() in your dialplan. Try dropping that in before you dial out. I have seen this so many times and Answer() has always fixed the issue. The magic number is 30 seconds.<br>
<br>Depending on if you use your CDRs for anything, especially billing, you may need to figure a way around that, since even if a call rings out, the CDR will reflect Answered.<br><br>-- <br>Thanks,<br>Steve Totaro <br>+18887771888 (Toll Free)<br>
+12409381212 (Cell)<br>+12024369784 (Skype)<br><br><br><br><div class="gmail_quote">On Fri, Nov 7, 2008 at 10:38 PM, Grey Man <span dir="ltr"><<a href="mailto:greymanvoip@gmail.com">greymanvoip@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">To get to the bottom of it I'd recommend determining why the ACKs are<br>
not getting through to Asterisk rather than trying to work around it.<br>
I'm actually suprised Asterisk terminates the call by default when it<br>
doesn't get the ACK to it's 200 Ok response that must be new for<br>
1.4.22 as I haven't seen that behaviour in earlier versions. In my<br>
opinion it's unwarranted behaviour, if Asterisk is getting RTP then it<br>
should leave the call up irrespective of whether it gets an ACK or<br>
not.<br>
<br>
>From the original SIP trace the ACK does not appear to be arriving at<br>
your Asterisk server at all. Try doing a packet trace on the network<br>
segment where the calling SIP agent is and see where it's trying to<br>
send the ACK to. My guess would be your firewall is incorrectly<br>
handling the SIP messages. Generally it's very bad news to use an ALG<br>
or firewall to mangle SIP packets as they almost always get it wrong.<br>
<br>
In your case there is a Record-Route header in the response so the ACK<br>
request should be being sent to that address. Perhaps your firewall is<br>
not correctly mangling that to allow the request to find its way back<br>
to your Asterisk server.<br>
<div class="Ih2E3d"><br>
Record-Route: <sip:<a href="http://216.82.224.202" target="_blank">216.82.224.202</a>;lr;ftag=as3ed791f3><br>
<br>
</div>Regards,<br>
<br>
Greyman.<br>
<div><div></div><div class="Wj3C7c"><br>
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