On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood <[email protected]> wrote: > On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby <[email protected]> > wrote: > > On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood <[email protected]> wrote: > >> > > My experience with Asterisk in the past has been with inbound analog > lines so that would make sense :) > > See if you spot anything weird here: > > 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.
-- Thanks, --Warren Selby http://www.selbytech.com
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