[Asterisk-Users] spa 3000 help

[Asterisk]

Hmm, I'm having real problems with the pstn side of things - I dial my 
number, and it is supposedly rnging, but the sipura stays quiet. If I dial a 
external number through * , then I get a "service is unavailable" which 
implies that the pstn is not present.

Does anyone have the settings for a uk pstn ? Tip/Ring voltage etc ?


----- Original Message ----- 
From: "Rich Adamson" <radamson at routers.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Thursday, October 07, 2004 2:22 PM
Subject: Re: [Asterisk-Users] spa 3000 help


>> Arrggghh. Tearing my hair out here.
>>
>> I'm trying to set up the spa3000 in the UK for my home, and want * to
>> control the dial plan
>>
>> I've googled to no avail. I've read the manual to no avail.
>>
>> Can someone, please let me know what the parameters is the spa and * are 
>> to
>>
>> a) receive a call from the pstn
>> b) make a call to the pstn from the phone attached
>>
>> I can make sip to sip calls (i.e. I can use xlite to call the phone, and 
>> the
>> phone to call xlite)
>
> For calls initiated from Line1, create a dialplan under the Line1 tab that 
> looks
> something like the following:
> (81xxx.<:@1.2.3.4;usr=3020;pwd=mysecret>|3xxx<:@gw1>|0<:@gw0>|[2-9]11<:@gw0>|xxxxxx.<:@gw0>)
> which means:
> - if the user dials 8-1-xxx-xxxx, the call is routed to asterisk
> - if the user dials 3xxx, route the call to gw1 (which is defined also as 
> asterisk)
> - if the user dials 0, route to the pstn port
> - if the user dials 911, 411, etc, route to the pstn
> - otherwise if no match, send the call to the pstn
>
> One item (of several) that aren't very clear from the documentation is the 
> use of
> gw0, gw1, etc, within the spa3k dialplans. For the line1 dialplan, gw0 
> defaults to
> the pstn port. Therefore in the above dialplan, 0<:@gw0 sends the call to 
> the pstn
> port. You'll notice two different ways in the above dialplan to send calls 
> to *.
>
> Within the spa3k, configure Line1 to register with asterisk and, under the 
> pstn tab,
> configure that interface to register with asterisk. These are two 
> completely different
> registrations and should have different User ID entries (eg, extn 1111 and 
> 2222).
> Once the two entries are properly entered, check using * cli with 'sip 
> show registry'
> to ensure the registrations are working as expected.
>
> Calls from asterisk to Line1 use a standard Dial(Sip...) dialplan within 
> asterisk.
> Calls routed from asterisk to the PSTN port use the standard Dial(Sip...) 
> entry as
> well, using the appropriate UserID for (eg, 1111 or 2222) entered in the 
> spa3k
> registration above.
>
> Due to the way I'm using the spa3k, I have all incoming pstn calls ring 
> the line1
> without passing through asterisk. So, not sure what parameters are used to 
> direct
> those calls to asterisk instead. My understanding is that others have done 
> this.
>
> Someone on this list noted that calls originating from asterisk and sent 
> out via the
> pstn interface are now failing. My implementation test of this about a 
> week ago
> suggests something changed within asterisk that precludes sending the 
> dialed number
> to the spa3k, but I've not had any time to trace the issue to identify the 
> root cause.
> It is entirely possible my asterisk config needs a little tweaking; not 
> sure yet.
>
> For my implementation, incoming pstn calls ring the phones attached to 
> line1. If
> asterisk sends a call to line1, I've configured the spa3k to provide 
> distinctive
> ringing to provide some indication where the incoming calls are coming 
> from. Works
> great.
>
> FWIW, I'm running cvs head from mid September along with the latest spa3k 
> firmware.
> I do have some echo issues (as others have) with the spa3k and apparently 
> Sipura
> is working on that (based on their release notes).
>
> Rich
>
>
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