asterisk users - Oct 2012 - as soon as Phone rings I'm disconnected yet phone rings two more times‏

One of the things I'm trying to do it to connect my 8x8 DTA 310 terminal
adapter onto my asterisk.  I have the 8x8 box connected to the Internet, and the
phone line connected to an fxo port on a Cisco router:

voice-port 0/2/0
 connection plar opx 5000
 caller-id enable

dial-peer voice 200 voip
 destination-pattern 5...
 session protocol sipv2
 session target sip-server
 codec g711ulaw
!         
sip-ua    
 sip-server ipv4:172.16.200.212     <------ Asterisk server

When I make a call from the PSTN to the 8x8 box, it does send ring back to the
asterisk server and the Digium phone does ring.  However, as soon as the phone
rings the call disconnects yet the actual phone, extension 5000, rings two times
before it hangs up, also.

The following output is what I see on the Asterisk console:

asterisk*CLI> 
  == Using SIP RTP CoS mark 5
[Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite: Call
from '' (172.16.200.1:65451) to extension '5000' rejected
because extension not found in context 'default'.
  == Using SIP RTP CoS mark 5
    -- Executing [5000 at pstn-incoming:1]
Dial("SIP/172.16.200.1-00000006", "SIP/5000,20|p") in new
stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/5000
    -- SIP/5000-00000007 is ringing
  == Spawn extension (pstn-incoming, 5000, 1) exited non-zero on
'SIP/172.16.200.1-00000006'

The 172.16.200.1 is my router.

sip.conf excerpt:

[5000]
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw
secret=cisco123
mailbox=5000 at phones

[172.16.200.1]
context=pstn-incoming
type=friend
host=172.16.200.1
dtmfmode=rfc2833
disallow=all
allow=ulaw

[phones]
exten => 5000,1,Dial(SIP/${EXTEN},20|p)
exten => 5000,n,Hangup

[pstn-incoming]
include=phones

Any help would be greatly appreciated,

Thanks,
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On Tue, Oct 23, 2012 at 7:54 PM, Mitchell Johnson
<mitch.johnson7 at gmail.com>wrote:
>
> One of the things I'm trying to do it to connect my 8x8 DTA 310
terminal
> adapter onto my asterisk.
>
What version of Asterisk are you using?

[Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352
handle_request_invite:
> Call from '' (172.16.200.1:65451) to extension '5000'
rejected because
> extension not found in context 'default'.
>
Did you mean to include this notice in your email? It indicates a dialplan
problem.

>     -- Executing [5000 at pstn-incoming:1]
Dial("SIP/172.16.200.1-00000006",
> "SIP/5000,20|p") in new stack
>
The pipe has been deprecated in more recent versions of Asterisk, make sure
this isn't related to your issue.

-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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