<div class="gmail_quote">On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood <span dir="ltr"><<a href="mailto:schmoe@gmail.com">schmoe@gmail.com</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><div></div><div class="h5">On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby <<a href="mailto:wcselby@selbytech.com">wcselby@selbytech.com</a>> wrote:<br>
> On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood <<a href="mailto:schmoe@gmail.com">schmoe@gmail.com</a>> wrote:<br>
>><br>
<br>
</div></div>My experience with Asterisk in the past has been with inbound analog<br>
lines so that would make sense :)<br>
<br>
See if you spot anything weird here:<br>
<br clear="all"></blockquote></div><br>Try adding "insecure=invite" to the DID_NUMBER peer, reload SIP and try your call again. By the way, it looks like your SIP provider has a built-in auto-failover to voicemail setup. You may want to get them to disable that once you get everything working on your end.<br>
<br>-- <br>Thanks,<br>--Warren Selby<br><a href="http://www.selbytech.com">http://www.selbytech.com</a><br>
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