Sonora Communications, Inc.

Application

Here we configure the SPA-3102 for use with Thirdlane (Asterisk) PBX.

We will configure the SPA-3102 as a SIP extension (ATA) and as an FXO (analog) gateway.

As an FXO (analog) gateway, will use the simplest IP-based call routing as opposed to registrations. No additional security will be configured, so this application is probably best kept behind your firewall.

We do not configure the SPA-3102 as a network gateway device. In fact, the LAN port will only be used for initial programming with a notebook computer, then will be left disconnected after that.

The PBX, phones and SPA-3102 will all be on the same network behind a firewall.

Interface

When connecting the SPA-3102, use the WAN interface to register to the PBX. The unit will apparently not register via the LAN interface.

Thirdlane PBX

SIP Trunk with IP Routing (no registration)

• Name
  • PSTN-1
• Description
  • Linksys SPA-3102
• Host
  • Your SPA-3102 IP address
• Direction
  • Both
• Context
  • From-Inside
• CODECs
  • G.711u
  • G.729
  • GSM
• DTMF
  • RFC2833
• NAT
  • no
• Can reinvite
  • no
• Other options:
  • incominglimit=1
  • outgoinglimit=1
  • call-limit=1
  • port=5061

Save and Reload.

SIP Extension

• Phone Model
  • none

Note the extension number and password for use programming the SPA-3102.

SPA-3102 Configuration

Default the SPA-3102 to factory defaults using a standard analog telephone plugged into the 'phone' port.? Disconnect the phone line from the 'line' port, if you have one plugged in.

Dial ****73738#
Then dial 1 to confirm

The WAN interface defaults to DHCP (client), which is usually what you want.

The LAN interface defaults to 192.168.0.1 where you can access the unit's web management interface.

The NAT and DHCP server functions default to on, but as you probably won't plug any devices into the LAN port (other than when programming the unit), that shouldn't matter.

http://192.168.0.1/admin/advanced

Click admin and advanced to see all of the configuration options.

Router Tab

WAN Setup

Enter the following for your location:

• Connection Type
  • Static IP
• Static IP
  • fixed IP address on your LAN
• Netmask
  • netmask of your LAN
• Gateway
  • your Internet router IP
  • may be your PBX
• Primary DNS
  • 8.8.8.8 (or your preferred DNS server)
• Secondary DNS
  • 8.8.4.4 (or your preferred DNS server)
• Primary NTP Server
  • pool.ntp.org
• Remote Management
  • Enable WAN Web Server

Submit All Changes.

Voice Tab

SIP

• RTP Packet Size
  • 0.020 (for G.711u)

Provisioning

• Provision Enable
  • no

Regional

• Control Timer Values
  • Interdigit Short Timer
    • 10
• Miscellaneous
  • Time Zone
    • your time zone
  • Daylight Saving Time Rule
    • in AZ, USA, we don't observe DST so I deleted the rule
  • DTMF Playback Length
    • set to .5 for better in-call touch tone response

Line1

• Line enable
  • yes
• NAT Mapping Enable
  • no
• SIP Port
  • 5060
• Proxy
  • IP of your PBX
• Register
  • yes
• Display Name
  • extention number
• User ID
  • extention number
• Password
  • password you set for this extention
• DTMF Tx Method
  • INFO
• Hook Flash Tx Method
  • INFO
• Dial Plan
  • The default dialplan is functional
    • You can just add a section after the '00|' to dial local extensions
      • '71xx' for 7100-series extensions
  • It may be better to match exactly as wildcards cause a delay in processing
  • You can terminate dialing with the '#' key
    • works like a 'Send' key on other phones
  • This dial plan allows 7 and 11 digit dialing only
    • (xxxxxxx|xxxxxxxxxxx)
  • This USA dial plan allows for faster dialing in most cases
    
    • ([2-9]xxxxxxS0|1[2-9]xxxxxxxxxS0|*xx|x.)
      • 7-digit numbers dialed immediately
      • 10-digit long distance numbers dialed immediately
      • dials 2-digit star codes
      • dials almost any other numeric sequence with a timeout

PSTN

• Proxy and Registration
  
  • Proxy
    • your PBX IP
  • Register
    • no
  • Make Call Without Reg
    • yes
  • Ans Call Without Reg
    • yes
• Subscriber Information
  • Display Name
    • PSTN
• Dial Plans
  • Dial Plan 1
    • (S0<:7101>)
      • forwards incoming analog calls to extension 7101 on the PBX
      • change '7101' to your desired destination extension
      • create a feature extension to send to an IVR
  • Dial Plan 2
    • (*x.|x.)
      • Outgoing
      • Dials most anything
• VoIP-To-PSTN Gateway Setup
  • Line 1 VoIP Caller DP
    • 2
  • VoIP Caller Default DP
    • 2

• PSTN-To-VoIP Gateway Setup
  • PSTN Ring Thru Line 1
    • no
  • PSTN CID For VoIP CID
    • yes
• FXO Timer Values
  • PSTN Answer Delay
    • 3
    • may have to adjust for accurate caller-ID detection
• International Control
  • FXO Port Impedence
    • 220+820||120nf
    • adjust for echo
  • SPA To PSTN Gain
    • 5
    • adjust for volume or echo
  • PSTN To SPA Gain
    • 5
    • adjust for voume or echo

Troubleshooting

• You might get in the habit of unplugging and re-plugging the power to the SPA-3102 frequently, or any time it behaves in a way you don't expect.
• Reset the SPA-3102 to factory defaults per above, and start over.

Asterisk sip.conf Example

[fxo-gateway]
qualify=no
nat=no
incominglimit=1
outgoinglimit=1
port=5061
;=description=Linksys SPA-3102
host=192.168.30.35
dtmfmode=rfc2833
context=from-inside
type=friend
call-limit=1
canreinvite=no
disallow=all
allow=ulaw
allow=g729
allow=gsm

[7120]
qualify=no
nat=no
pickupgroup=1
callerid= SPA-3102 <7120>
context=from-inside
canreinvite=no
vmexten=7120
secret=MwveKq
username=7120
host=dynamic
callgroup=1
subscribecontext=local-extensions
dtmfmode=rfc2833
type=friend
mailbox=7120
disallow=all
allow=ulaw
allow=g729
allow=gsm