<br><br><div class="gmail_quote">On Tue, Mar 5, 2013 at 2:32 PM, Hose <span dir="ltr"><<a href="mailto:hose+asterisk@bluemaggottowel.com" target="_blank">hose+asterisk@bluemaggottowel.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
We have an asterisk frontend terminating all our SIP phones to, and an<br>
asterisk backend with a wildcard PRI card in it connecting to the PTSN.<br>
The frontend handles 99% of dialplan logic and just hands off anything<br>
outgoing to the backend via IAX2, which dials out on one of the open<br>
channels.<br></blockquote><div><br></div><div>IAX is buggy. We've never seen a reliable system using it. We've given up on it. I'd try SIP. Easy to do, no real reason not to.</div><div><br></div><div>Check all of the networking involved. Leave a ping test running between the systems constantly, then see if it dropped packets when you get a dropped call.</div>
<div><br></div></div><div><br></div>-- <br><div>Carlos Alvarez</div><div>TelEvolve</div><div>602-889-3003</div><div><br></div>