Sipura 3000 configuration with Asterisk

In this article I will try to explain that how to configure Sipura 3000 PSTN with Asterisk.

First of all let’s analyize a little background of Sipura devices, After that maybe we will try to configure Sipura spa 3000 gateway with Asterisk.spa_3000.jpg

Sipura SPA-3000 Analogue Phone Adapter
The SPA-3000 continues to deliver on Sipura Technology’s mission to provide market leading, best-in-class VoIP end points providing freedom and opportunity to service providers and end users.

The SPA-3000 features VoIP adapter functionality found in the SPA-2000 and SPA-1000/1 with the additional benefit of an integral connection for legacy phone network� applications. SPA-3000 users will be able to leverage their broadband phone service connections more than ever by automatically routing local calls from cell phones and land lines to a VoIP service provider and vice versa.

Configuration:

I have already written some notes on Sipura device but This one a little long – hopefully not too boring! Very Happy

Asterisk side configuration:
In your Sip.conf file you need something along the lines of…

Code:

; This ones for the analogue phone
[LinkSys1] ; used to identify your spa within extentions.conf
context=nicepeople ; the context for the device to sit in.
cancallforward=yes
dtmfmode=rfc2833
type=friend
host=dynamic ; I haven't got fixed ip addresses
username=LinkSys1 ; used back in the spa
secret=password
authuser=LinkSys1
callerid=AnaloguePhone
canreinvite=no ; Let * be the middle man in all the calls.

; This one is for the ‘line’ connection to the outside world
[LinkSysOut] ; used to id this within your extentions.conf
context=bell ; this is the context that will get jumped to when a call comes in on this. (I’ve identified it separate here to the phone to keep dial plan privileges separate)
cancallforward=yes
dtmfmode=rfc2833
type=friend
host=dynamic
username=LinkSysOut ; used in the spa
secret=password
authuser=LinkSysOut
callerid=Line1
Nat=no
canreinvite=no ; Let * be the middle man in all the calls.

Obviously passwords need to be set – things like callerids can be changed – canreinvite could be set to yes but I have an Internet connection so the asterisk box needs to stay in the loop to get through my firewall.

OK that’s the sip.conf – a quick note on my extentions.conf in this example.
context=nicepeople is where I stick my phones (OK – devices) that I use – each can call each other and access the outside world etc… Normal extentions.conf stuff really.
context=bell is used to hold my analogue lines (ma-bell:) I’ve got a few inbound macros that run to screen incoming calls so all my inbound stuff from the outside world goes through the same context.

Sipura side configuration:

Your SPA is where the fun starts Smile
Now You may have to try things a few times and get back to me if it doesn’t work and I’ll look a bit deeper.

First off – log-in as administrator and select advanced view. If I don’t talk about a setting – its the default.

OK going in tab order

System
The easy one, set your IP address/ dns/ gateway to fit within your network on the system tab.

Regional
Caller ID Method = Bellcore(N.Amer,China) – Although I’m in the UK, on an NTL analogue line, they use Bell core standards for caller line id.

Line1
Line Enable = yes
Sip Port = 5060
Proxy = ip address of your asterisk box
Register = Yes

The next bit (still on the Line1 tab) correspond to the part of your sip.conf file we were setting above.
Display Name = LinkSys1
User ID = LinkSys1
Password = password
Auto PSTN Fallback = Yes

PSTN Line
Line Enable = yes
Sip Port = 5061
Proxy = ip address of your asterisk box
Register = Yes

This is similar to the Line1 tab and also ties into your sip.conf file. The difference this time is this is for the actual analogue line (the one that you plug into the wall not the phone!).
Display Name = LinkSysOut
User ID = LinkSysOut
Password = password

Dialplan 8 = S0< :s> ; this is key later on. I’ve picked dialplan 8 but you can use any spare – as long as you make a note of the dialplan you use. The < :s> basically says go to the first or start extension within the context that we set in sip.conf (I set this to [bell])

VoIP-To-PSTN Gateway Enable = yes (you do want to dial out I guess.)
PSTN-To-VoIP Gateway Enable = yes (and we do want inbound calls)
PSTN CID For VoIP CID = yes passes the caller id.
PSTN Caller Default DP = 8 – Remember I said you can pick your own dial plan above – this is where you tell it which one you have set the S0< :s> in.

You may have to play with these to get the spa to detect call ending…
Detect CPC = yes
Detect Polarity Reversal = yes This made the biggest difference with NTL – as soon as the line drops this picks it straight away! I used to use a x100p clone card which took a while to detect disconnection – this is right on the button!
Detect Disconnect Tone = yes

Sorry if this is a bit long winded – but I had the problem, when I was looking into this originally, where others would post the config but not really explain what was doing what – hopefully this does a little better

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