[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 21:49:37 CDT 2010

On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood <schmoe at gmail.com> wrote:

> I don't see any
> On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby <wcselby at selbytech.com>
> wrote:
> >
> > You don't have any extensions in your default context that match the
> > extension that your sip peer is dialing in on.  's' is not a default
> > extension for SIP...try using _X., and see what you get.  Bump up the CLI
> > (core set verbose 10) and then repost a failed called attempt.  Some SIP
> > providers also use a + symbol in front of their inbound calls, so you may
> > need to use _+X., instead.
> I don't see any call attempt/logs when I bump up the verbosity, and
> when I check my verbose logs I show:
The next step would be to enable sip debug on the peer you're trying to
receive calls from (sip set debug peer PEERNAME or sip set debug ip
IPADDRESS).  Then send another call inbound and see what's happening.  As
far as the 's' extension, that's the "start" extension, it's used when no
other extension information is presented.  Pretty much every SIP peer I've
ever seen presents an extension when entering a context, and thus the 's'
extension doesn't catch it.  I've typically only seen 's' used in Macros and
with inbound analog lines.

--Warren Selby
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