Configuring Sipura SPA-3000 as trunk within Asterisk VoIP PBX Server

by JRKLEIN on OCTOBER 15, 2006 · 2 COMMENTS

This article describes how I successfully configured the Sipura SPA-3000 (fw 2.0.13) for use as a single line inbound/outbound trunk within Asterisk at Home (asterisk 1.2.1). Unlike the other examples I found, this configuration is fairly simple and does NOT require configuration of special extensions, etc. This configuration should be fairly secure, but any suggestions and/or feedback are very welcome!

When incoming calls are received by the SPA-3000, they are forwarded to the Asterisk PBX with CALLER ID information and can be routed like any other POTS trunk (ie: as per Incoming Calls config and/or Inbound Routing config by CID). When outgoing calls are placed through the SPA-3000, this device dials the number and connects the call. The person making the call WILL hear the DTMF tones (aka touch tones) that are dialed by the SPA-3000 just before the call is connected. I have not been able to find a way of preventing this (yet).

Configuring Trunk within Asterisk PBX using AMP

General Settings

1.Outbound Caller ID: (leave blank - cannot be used by POTS line)

2.Maximum Channels: 1 (required - see note below)

NOTE: Each SPA-3000 supports a single channel. You need to setup multiple trunks for multiple SPA-3000 devices.

Outgoing Dial Rules

Dial Rules:
1+NXXNXXXXXX        ; prefix 10 digit dialing with "1"
1NXXNXXXXXX     ; allow all 11 digit dialing as-is
NXXXXXX         ; allow all 7 digit dialing as-is

Outgoing Settings

Trunk Name: pstn_spa01
 
Peer Details:
auth=md5
context=from-pstn
dtmfmode=inband
fromuser=asterisk
host=10.10.10.21    ; IP address of SPA device
insecure=very
nat=yes         ; omit if no NAT exists between PBX and SPA
port=5061
secret=012345678901
type=peer
username=asterisk

Incoming Settings

User Context: spa01
 
User Details:
allow=ulaw
context=from-pstn
disallow=all
dtmfmode=inband
host=10.10.10.21    ; IP address of SPA device
insecure=very
nat=yes         ; omit if no NAT exists between PBX and SPA
secret=KzBTALezmG1a
type=friend

Registration

1.Register String: ; omit - not necessary to register w/ SPA device?

Configuring Outbound Routing within Asterisk PBX using AMP

Add Route

Route Name:     ; user preference, avoid special characters here?
pstnspa1
 
Dial Patterns:      ; dial 5 plus 11 digit, 10 digit, and 7 digit numbers
; omit each "5|" to use trunk without dialing prefix
5|1NXXNXXXXXX       ; accept 5 + 11 digit dialing
5|NXXNXXXXXX        ; accept 5 + 10 digit dialing
5|NXXXXXX       ; accept 5 + 7 digit dialing
 
Trunk Sequence:     ; add each available SPA-3000 trunk
SIP/pstn_spa01
SIP/pstn_spa02
SIP/pstn_spa03

Configuring the Sipura SPA-3000

The following example only illustrates changes to default settings. Start by performing a factory reset of your SPA-3000. Connect a handset to the PHONE jack on the SPA-3000 and dial “****” to access the configuration menu, then dial “73738#” (aka “RESET#”) to perform a factory reset.

Login to the web interface of your SPA-3000, click “Admin”, then click “Advanced”. Configuration changes for each tab/page are shown below.

SYSTEM

USER PASSWORD:      secretpwd       ; secures the SPA web interface
; username 'user' or 'admin'?
 
DHCP:           no          ; recommend static ip address
STATIC IP:      10.10.10.21
NETMASK:        255.255.255.240
GATEWAY:        10.10.10.30
 
HOSTNAME:       voip-spa1       ; optional
DOMAIN:         example.net     ; optional
PRIMARY DNS:        10.10.10.2      ; optional
SECONDARY DNS:      10.10.10.3      ; optional
PRI NTP:        ntp1.example.net    ; optional
SEC NTP:        ntp2.example.net    ; optional

SIP

1.RTP Packet Size: 0.020 ; improves sound quality (was 0.030)?

REGIONAL

1.TIME ZONE: GMT-05:00 ; Central Time Zone

PSTN LINE

NAT Mapping Enable: yes ; only change if NAT exists between PBX and SPA
NAT Keep Alive Enable:  yes ; only change if NAT exists between PBX and SPA
 
PROXY:          10.10.10.24 ; IP address of Asterisk PBX
USE OUTBOUND PROXY: yes
REGISTER:       no
REGISTER EXPIRES:   3600
MAKE CALL W/O REG:  yes
ANSW CALL W/O REG:  yes
 
DISPLAY NAME:               ; leave blank
USER ID:        3501        ; optional?
PASSWORD:               ; leave blank
 
DTMF Process INFO:  Yes     ; default value
DTMF Process AVT:   No      ; resolve issues with DTMF
DTMF Tx Method:     Auto        ; default value
 
DIAL PLAN 8:        (S0&lt;:s@10.10.10.24:5060&gt;)
; forwards incoming PSTN calls to PBX
; resolve issues with DTMF
 
VOIP-TO-PSTN GW ENABLE: yes
VOIP CALL AUTH METHOD:  http digest
ONE STAGE DIALING:  yes
LINE1 VOIP CALLER DP:   none
VOIP CALLER DEFAULT DP: none
LINE1 FALLBACK DP:  none
 
VOIP USER 1 AUTH ID:    asterisk
VOIP USER 1 DP:     none
VOIP USER 1 PASSWORD:   012345678901
 
PSTN-TO-VOIP GW ENABLE: yes
PSTN CALL AUTH METHOD:  none
PSTN RING THRU LINE 1:  no      ; incoming calls do not ring LINE1
PSTN CID FOR VOIP CID:  yes
PSTN CALLER DEFAULT DP: 8
 
PSTN ANSWER DELAY:  5       ; answer incoming PSTN call in X sec
; need to allow time for CALLER ID
; if no CID, you can safely set to 0
; was set to 16