[Asterisk-Users] Sipura 3000 FXO with Asterisk

[Razza]

All,
Further to my note below, I now have incoming working - yipee! (and seem
to have identified a problem with the G711A codec in the latest sipura
firmware - although need to do some checking). This box sounds great
compared to the echo ridden FXO and gives me an FXS for very little more
cash.

I now have a really strange issue for outgoing calls, everything seems
ok including the SIP messages (i.e. <dialled number>@<sipura ext>) but I
am always getting through to a wrong number (fortunately I'm doing this
on a Sunday and it's a business number so I'm just getting their answer
machine).

I have included excepts from my test extensions.conf and sip.conf files,
could someone please confirm these are ok (for my own sanity)? The other
strange thing is the sipura info tab tells me 'Last Called PSTN Number'
is correct. 

I assume I have got something very wrong with the sipura config,
although have not changed anything - so assistance on the sipura would
be greatly appreciated.

-------------------------
*** extensions.conf ****
-------------------------
[general]
static=yes
writeprotect=no

[globals]
CC=UK
CONSOLE=Console/dsp

[sip_home]
exten => 100,1,SETCIDNUM(${CALLERIDNUM:1}) ; strips leading character
added to CLI by the SPA3K to frig no answer issue

exten => 100,2,Dial(SIP/budget1,25,tr)

exten => _0X.,1,Dial(SIP/${EXTEN:0}@101,60,r)

exten => 105,1,Dial(SIP/budget1,20tr)


-------------------------
******* sip.conf *******
-------------------------
[general]
%< ------ SNIP ------- >%

[101]
;PSTN
type=friend
regexten=101
username=983
secret=razza
context=sip_home
port=5080
host=dynamic
nat=no
canreinvite=no
disallow=all
;allow=alaw
allow=ulaw

[budget1]
type=friend
regexten=105
username=budget1
secret=razza
context=sip_home
callerid="Kitchen" <105>
host=dynamic
nat=no
;canreinvite=no
disallow=all
;allow=alaw
allow=ulaw

Regards,
Ray


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Razza
Sent: 16 April 2005 00:21
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Sipura 3000 FXO with Asterisk


Pete wrote: 
> The comments about it being an ugly "hack" arent really correct.  The
Sipura is really built > for standalone useage wiht a sip provider
however it does work well with asterisk.
>
> Follow this thread
>
>http://voxilla.com/forum-viewtopic-t-1335.html
>
>it works and it works **VERY** well :-)

>Pete

Help!!!
I have spent the whole day trying to get this to work and simply cant,
I'm aware the instructions are very simple but there is no sip traffic
generated to the * server from the SPA-3000 when I call my PSTN number
(outgoing from sip phone to spa-3000 through * is fine) - are there
other settings I am missing?

As I am in the UK I have also changed the line impedences according to
http://www.sinet.bt.com/351v4p2.pdf and have changed the 'Caller ID
Method' (in regional tab) to 'ETSI FSK WithPR (UK)' but still nothing.

Anyone able to send me screen dumps of their config or advise?

Ray.
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