Installing a Linksys SPA3102 with FreePBX

Friday, Oct 3, 2008

After doing a clean install of Elastix 1.3 on a VMWare Server virtual machine (8GB hard drive, 380MB RAM), here’s a record of what I did (following mostly this guide):

Reset the SPA3102

Factory reset the SPA3102 (remove the Internet and Line cables, dial **** and 73738# 1)

Set the SPA3102 to obtain an IP address through DHCP (****, 101# 1# 1)

Enable web server (7932# 1# 1)

(Operation codes to administer the SPA3102 through the phone can be found in the Linksys SPA3102 user guide)

Elastix settings (using FreePBX)

Add a SIP trunk.

General Settings

Outbound Caller ID: “Jorge” <5551234>
Maximum Channels: 1
Dial Rules: XX.
Trunk Name: etb

Peer Details

disallow=all
allow=ulaw
canreinvite=no
context=from-trunk
dtmfmode=rfc2833
host=dynamic
incominglimit=1
nat=never
port=5061
qualify=yes
type=friend
username=etb
secret=*******

User Context: leave empty

User Details: leave empty

Register String: leave empty

Outbound routes

Create one or more outbound routes for local and long distance calls, setting the Trunk sequence to be the SIP trunk just created. A typical Dial pattern is 1|NXXXXXX, which removes the leading area code (1) for local calls. Remember to add an “emergency” route with common three-digit emergency numbers.

Inbound route

DID number: 5551234 (has to match the SPA3102 Dial plan setting)
Destination: extension 101

Extensions

secret: ****
dtmfmode: rfc5833
canreinvite: no
context: from-internal
host: dynamic
type: friend
nat: no
port: 5060
qualify: yes
dial: SIP/101
mailbox: 101@device
call-limit: 4 (?)

SPA3102 Settings

SIP

RTP Packet size: 0.020

PSTN Line

SIP Settings
SIP Port: 5061

Proxy and Registration
Proxy: 192.168.**.* (IP of asterisk server)
Make Call Without Reg: yes
Ans Call Without Reg: yes
Register Expires: 300

Subscriber Information
Display Name: Entrante ETB
User ID: etb
Password: **** (the same as in SIP Trunk)

Audio Configuration
DTMF Process INFO: Yes
DTMF Process AVT: Yes
DTMF Tx Method: Auto

Dial Plans
Dial Plan 2:(S0<:5551234>) (the same as in Inbound route)
All others: (xx.)

VoIP-To-PSTN Gateway Setup
VoIP-To-PSTN Gateway Enable: yes
VoIP Caller Auth Method: None
VoIP PIN Max Retry: 3
One Stage Dialing: Yes
Line 1 VoIP Caller DP: none
VoIP Caller Default DP: none
Line 1 Fallback DP: none

PSTN-To-VoIP Gateway Setup
PSTN Ring Thru Line 1:no
PSTN CID For VoIP CID:yes
PSTN Caller Default DP:2

FXO Timer Values (sec)
PSTN Answer Delay:3

International Control
SPA To PSTN Gain:2
PSTN To SPA Gain:2

Regional tab

Left unchanged (Elastix without tears changes several parameters to Australian settings; localized codes for Colombia can be found here)

Delete all vertical services activation codes

Line 1 tab

Proxy: 192.168.**.* (IP of asterisk server)
Register Expires: 60
Display Name: Wireless
User ID: 101 (extension number)
Password: ***** (same as in Extensions)

DTMF Tx Method: Auto (INFO ?)
Hook Flash Tx Method: None (INFO ?)
Dial Plan: ([1-9]xxx.|03xxx.|07xxx.|007xxx.|*xx)

User 1 tab

Default Ring: 1 (3 ?)
Default CWT: 1 (8 ?)