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

Warren Selby wcselby at selbytech.com
Wed Aug 4 22:41:21 CDT 2010

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.

--Warren Selby
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100804/ce42be65/attachment.htm 

More information about the asterisk-users mailing list