[Asterisk-Users] Sipura 3000 FXO with Asterisk

[Ed Greenberg]

Hi Razza,

I don't know what country you are in, or what your country's telephone 
numbers look like, but it seems from your dialplan that if you dial an 
outside number it needs to start with 0X.

So if you dial 012345, the Sipura will dial 012345 on the fxo port.

If your line needs to dial 12345, you should use ${EXTEN:1} to drop the 
zero off the beginning.

I recommend that you run the console with verbose on (asterisk -rvvvvv) and 
watch to see what is actually being dialed on the Sipura.

Best,
</edg>

If that is not the problem,

--On Saturday, April 16, 2005 11:50 AM +0100 Razza 
<rjames31 at btopenworld.com> wrote:

> 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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