[Asterisk-Users] Sipura 3000 FXO

Benjamin on Asterisk Mailing Lists benjk.on.asterisk.ml at gmail.com
Sat Oct 2 02:38:03 MST 2004


On Sat, 02 Oct 2004 00:08:34 -0500, denon <denon at denon.cx> wrote:

> If anyone has one that is working, on any firmware, I'd appreciate if
> they'd try a *67 and see if it works for them, though.

I borrowed an SPA-3000 in order to a) see if it works on the Japanese
PSTN network and b) write a setup assistant for Asterisk on MacOSX so
users can GUI-configure Asterisk to use it as an FXO gateway for both
incoming and outgoing PSTN calls.

I have outgoing PSTN calls (Asterisk--->SPA--->PSTN) working but not
incoming, not yet at least.

Let me first list the setup I use for outgoing calls and then I will
talk about how I think you could get your "*67" inserted in the
outbound DTMF sequence of the SPA when it dials.

ASTERISK SIDE:

in /etc/asterisk/sip.conf ...

[fxogw1]
type=friend
port=5061
auth=md5
secret=blah
host=dynamic
context=incoming

(you may still fine tune this with other parameters, like
nat/reinvite, qualify, codecs, callerid etc)

NOTE: I do not recommend the use of type=friend for anything other
than testing. Once I have both incoming and outgoing working I will
split this into two entries, one type=user and one type=friend, which
is how it should be done.

in /etc/asterisk/extensions.conf

[pstn-out]
;
exten => _9X.,1,NoOp(Outgoing call to PSTN ${EXTEN:1})
exten => _9X.,2,Dial(SIP/${EXTEN:1}@fxogw1)
exten => _9X.,3,Hangup

(Don't forget to include context "pstn-out" in whatever context it is
your phones are assigned to)


SPA-3000 SIDE:

in the "PSTN Line" tab ...

Line enable: yes

SIP port: 5061
Proxy: 192.168.100.20	(Note: this is the IP address of the Asterisk server)
Use outbound Proxy: no
Register: yes
User ID: fxogw1
Auth ID: fxogw1
Use Auth ID: no
Password: blah
Preferred Codec: G711u		(or whatever other codec you may want to use)
DTMF tx method: INFO
VoIP-to-PSTN-Gateway enable: yes
VoIP caller auth method: none
One stage dialling: yes
VoIP caller default Dialplan: none
VoIP answer delay: 0

I also had to set FXO port impedance to "Global" for use here in
Japan, so if you are not in the US, you may have to change this
setting (just try out all the choices until you succeed).


The above works fine for dialling out from Asterisk through the SPA to
a PSTN line. However, I had to upgrade the firmware of the SPA-3000 to
2.0.10 in order to get it to dial out sliently. With the original
firmware it gave an audio feedback of the PSTN dialtone and the DTMF
of its dialling out.

Now, if you want the SPA to dial a DTMF sequence with a prepending
"*67" sequence or anything else that contains * or #, I don't think
you should insert that on the Asterisk side. I think you should let
the SPA insert this and you can use the SPA's dialplan feature to do
so.

I have only just started to play the the dialplan settings on the SPA,
so I can't give you the sequence which would insert *67, but if you
take a look at the SPA's User's Guide (download from Sipura's website)
there is a section on how to define dialplans and it supports
replacement and insertion strings.

something like 

<:*67><:@gw0>

or maybe

<:*67>xx.<:@gw0>

should in theory prepend "*67" to anyhing you dialled from Asterisk
then send it to the PSTN line connected to the FXO port (designated by
"gw0")

If you want to selectively dial with and without the "*67" sequence,
then you could set up multiple dialplans and use a numeric only prefix
from Asterisk, ie 00067 for inserting "*67" or you could set up
multiple VoIP users (accounts) on the Sipura and send them to
different dialplans.

Anyway, I am still struggling with this Sipura box myself, so you will
have to test this to see if the idea behind my suggestion is right.

If you get incoming calls working, I'd appreciate if you could share
your setup. When I try to call in on the FXO port, the SPA picks up
the line and gives me a fast busy signal. It doesn't even make an
attempt to send the call anywhere. The suggestions Sipura support
emailed me don't work either.

rgds
benjk

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

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.



More information about the asterisk-users mailing list