[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

[Warren Selby]

On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood <schmoe at gmail.com> wrote:

> On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby <wcselby at selbytech.com>
> wrote:
> > On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood <schmoe at gmail.com> 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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