In the past few weeks I worked converting our home and work telephone system to VoIP. Here you can see the schematics:

I’ve used the following VoIP adapters: Linksys SPA-3000, Linksys PAP2T, Linksys SPA-1001 (all from eBay of course). I have separate Skype users for home and the shop (skype-otthon, skype-uzlet) with a country-wide landline subscription (it cost €40 for a whole year), so Skype is used for every outgoing call to landlines. We kept our PSTN lines, everyone knows our shop’s number, so it couldn’t be changed to a Skype number and at home we can keep our PSTN line without subscription (outgoing calls will cost a lot if we use it, but of course we won’t).

My PBX is Asterisk, it runs on my home Linux server. You can find information about using Skype with Asterisk here.

A few pictures

This is the setup of our phone in the yard (muhely). This phone is hooked to an SPA-1001 and a TP-Link WiFi AP/router and this connects to my Ubiquity LocoStation Nano WiFi AP on the peach tree:

Networking stuff on my room’s wall, from left to right: the two VoIP adapters, DSL modem, 2 switches and a UPS:

PAP2T, SPA-3000, TP-Link and SMC switches in my Father’s office in our shop:

Linksys SPA-3000 configuration

I only write here the settings that differ from the default settings.

SIP tab
RTP Packet Size: 0.020

Regional tab
Ring Waveform: Sinusoid
Ring Voltage: 85
Ring Frequency: 25
Interdigit Long Timer: 4
Interdigit Short Timer: 2
Blind Transfer Code: ##
Time Zone: GMT+01:00
Caller ID Method: ETSI FSK
FXS Port Power Limit: 1

Line 1 tab
Network Jitter Level: low
Jitter Buffer Adjustment: disable
Proxy: 192.168.1.1
User ID: …
Password: …
DTMF Tx Method: AVT
Dial Plan: (1xx<:@gw0>|<9:><:@gw0>|#S0L2xx|1[01]S0x|1xxx|[23456]xxxxx|0[06]xxxxxxx.|[78]xx.)
Emergency Number: 1[01]x

PSTN Line tab
Network Jitter Level: low
Jitter Buffer Adjustment: disable
Proxy: 192.168.1.1
User ID: …
Password: …
DTMF Tx Method: AVT
Dial Plan 1: (S0<:pstn-otthon>)
VoIP-To-PSTN Gateway Enable: yes
Line 1 VoIP Caller DP: 2
PSTN-To-VoIP Gateway Enable: yes
PSTN Ring Thru Line 1: no
PSTN CID For VoIP CID: yes
PSTN Caller Default DP: 1
FXO Timer Values (sec): everything is 0, except: PSTN Hook Flash Len: 0.25, VoIP PIN Digit Timeout: 10, PSTN PIN Digit Timeout: 10, PSTN Ring Timeout: 3, PSTN Dial Digit Len: .5/.1
Disconnect Tone: 425@-20,425@-20;2(0.3/0.3/1)
On-Hook Speed: 3 ms (ETSI)

User 1 tab
VMWI Ring Policy: New VM Arrives

Linksys PAP2T configuration

SIP tab
RTP Packet Size: 0.020

Regional tab
Ring Waveform: Sinusoid
Ring Voltage: 85
Ring Frequency: 25
Interdigit Long Timer: 4
Interdigit Short Timer: 2
Blind Transfer Code: ##
Time Zone: GMT+01:00
Caller ID Method: ETSI FSK
FXS Port Power Limit: 1

Line 1 tab
Network Jitter Level: low
Jitter Buffer Adjustment: disable
Proxy: 192.168.1.1
User ID: …
Password: …
DTMF Tx Method: AVT
Dial Plan: (#S0L2xx|1[01]S0x|1xxx|[23456]xxxxx|0[06]xxxxxxx.|[78]xx.|9)
Emergency Number: 1[01]x

Line 2 tab
Network Jitter Level: low
Jitter Buffer Adjustment: disable
Proxy: 192.168.1.1
User ID: …
Password: …
DTMF Tx Method: AVT
Dial Plan: (#S0L2xx|1[01]S0x|1xxx|[23456]xxxxx|0[06]xxxxxxx.|[78]xx.|9)
Emergency Number: 1[01]x

User 1 tab
VMWI Ring Policy: New VM Arrives

User 2 tab
VMWI Ring Policy: New VM Arrives

Linksys SPA-1001 configuration

SIP tab
RTP Packet Size: 0.020

Regional tab
Ring Waveform: Sinusoid
Ring Voltage: 85
Ring Frequency: 25
Interdigit Long Timer: 4
Interdigit Short Timer: 2
Blind Transfer Code: ##
Time Zone: GMT+01:00
Caller ID Method: ETSI FSK
FXS Port Power Limit: 1

Phone tab
Line 2 Select Code:

Line 1 tab
Network Jitter Level: low
Jitter Buffer Adjustment: disable
Proxy: 192.168.1.1
User ID: …
Password: …
DTMF Tx Method: AVT
Dial Plan: (#S0L2xx|1[01]S0x|1xxx|[23456]xxxxx|0[06]xxxxxxx.|[78]xx.|9)
Emergency Number: 1[01]x

Line 2 tab
Line Enable: no

User 1 tab
VMWI Ring Policy: New VM Arrives

If you have an adapter behind NAT

Follow this guide.

Asterisk configuration

http.conf
I switched off HTTP interface, because I don’t need it. Asterisk console is fine for me (asterisk -r -d -v).

indications.conf
I’ve set the country to hu.

manager.conf
I’ve disabled Asterisk call management support, I don’t need it.

sip.conf
Here’s a config for an SPA-3000 PSTN line and a regular SIP line:

[pstn-otthon]
type=friend
secret=…
host=dynamic
callerid=”PSTN Otthon” <481365>
canreinvite=no
call-limit=1
context=otthon
#qualify=yes

[alsoszint]
type=friend
secret=…
host=dynamic
callerid=”Otthon also szint” <#01>
canreinvite=no
call-limit=1
context=otthon
#qualify=yes

Config for Skype stsProxy:

[skypeotthon]
type=friend
secret=skypeotthon
host=dynamic
canreinvite=no
context=otthon
#qualify=yes

features.conf
I’m only using blindxfer (call transfer):

[featuremap]
blindxfer => ##

extensions.conf
Download here.