asterisk users - Oct 2004 - [OT] Sipura-3000 - Immediate hangup on inbound PSTN calls

My apologies for the off-topic post ...

No matter what settings I try, when I dial in to the SPA-3000 on the
PSTN line, it picks up the call and immediately gives me a fast busy
tone then hangs up. The info tab says under PSTN Line status:

Last PSTN Disconnect Reason: PSTN Disconnect Tone

which seems to indicate that the SPA thinks the caller has hung up.

Since I am in Japan, it is possible that the SPA simply doesn't work
here. Or perhaps, some of the FXO parameters need to be changed. So,
my question is this: Has anybody experienced similar problems with an
SPA outside of the US and if so, did anybody find a fix/workaround?

thanks
rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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> My apologies for the off-topic post ...
> 
> No matter what settings I try, when I dial in to the SPA-3000 on the
> PSTN line, it picks up the call and immediately gives me a fast busy
> tone then hangs up. The info tab says under PSTN Line status:
> 
> Last PSTN Disconnect Reason: PSTN Disconnect Tone
> 
> which seems to indicate that the SPA thinks the caller has hung up.
> 
> Since I am in Japan, it is possible that the SPA simply doesn't work
> here. Or perhaps, some of the FXO parameters need to be changed. So,
> my question is this: Has anybody experienced similar problems with an
> SPA outside of the US and if so, did anybody find a fix/workaround?
Just seen the same thing here this morning (in the US). I had tested
outgoing pstn calls from * a couple of weeks ago and it worked fine.
Now it does not work. However, an ethereal trace indicates * is not
sending a dial string to the spa3k. Is that due to changes that occurred
with *, or, spa3k firmware upgrade that happened at roughly the same 
time (here). I don't know at this point. Without digging into it, I'd
have to guess (with absolutely no facts to back it up) that changes
with the sip protocol handling within * has something to do with it.

Outgoing calls via the spa3k is not our primary dialplan, so not sure
when it broke.

Rich

> Can you post spa 3000 config for inbound fxo to asterisk?
<http://rip.psg.com/~randy/spa3k.html>

and the sip.conf snippet

    [biwa-pstn-in]
    type=peer
    auth=md5
    secret=haha
    username=asterisk
    fromuser=asterisk
    host=spa3k.foo.edu
    port=5061
    dtmfmode=rfc2833
    nat=no
    canreinvite=yes
    context=spa3k-pstn-in

and the extensions.conf snippet

    [spa3k-pstn-in]
    exten => _X.,1,SetVar(areacode=666)
    exten => _X.,2,SetVar(mailbox=42)
    exten => _X.,3,GoTo(ext-common,s,1)

    [ext-common]
    exten => s,1,Background(zz-in-who-do-you-want)
    exten => i,1,Hangup()
    exten => t,1,GoTo(ext-common,s,1)
    include => speeddials
    include => extensions
    include => conferences
    include => applications

you might ask why the two contexts with goto.  the answer is
that without that it did not work because the first had to
consume the spa3k's extension from the sip connection.

hope this helps someone.  i tried to get the anonymization
correct so it would not make it hard to understand.

randy