[Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems

[Dan Perik]

I just got my shiny new Grandstream HandyTone-488 today.  My goal is to
use it to allow incoming/outgoing calls to PSTN using my normal ole'
phone as usual.  I will be switching over to using BroadVoice as my main
phone #, but want that to be as seemless of a switchover as possible
(for the wife and kids, and for people needing to call us).

I've got the following working:

FXS -> * ( and then -> BroadVoice )
( BroadVoice -> ) * -> FXS
FXO -> * ( and then -> FXS )

I don't have this working:
( FXS -> ) * -> FXO

In other words, I can't seem to call out on my PSTN line from Asterisk.

Here's a snippet from sip.conf:
[gs1-FXO]
type=friend
context=default
host=dynamic
username=gs1-FXO
secret=<mysecret>
nat=no
canreinvite=yes
dtmfmode=info
incominglimit=1
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g729

Here's a snippet from extensions.conf:
[gs1-fxo-out]
exten => _8.,1,Dial(SIP/${EXTEN:1}@gs1-FXO)

So when I dial, say 85429411, I would expect it to dial 5429411 out on
the PSTN line. I end up not getting any tone or other audio out of the
handset.  But, using another phone directly connected to the PSTN, I
find that the Grandstream has taken the line off hook, but not dialed
any digits.  I get this in my * log when I dial 85429411.

    -- Executing Dial("SIP/gs1-FXS-9041", "SIP/5429411 at gs1-FXO") in new
stack
    -- Called 5429411 at gs1-FXO
    -- SIP/gs1-FXO-877b is ringing
    -- SIP/gs1-FXO-877b answered SIP/gs1-FXS-9041
    -- Attempting native bridge of SIP/gs1-FXS-9041 and SIP/gs1-FXO-877b
  == Spawn extension (outgoing-ok, 85429411, 1) exited non-zero on
'SIP/gs1-FXS-9041'

I know the Handy-Tone 488 is a new device, so there may be some quirks
to it.  But I would think it _should_ work.

Any suggestions?

Thanks!
Dan