Asterisk is now working with the SPA3102!! :-)
I thought I’ll share my struggle here (most of the struggle was with figuring out the much poorly documented SPA3102 configuration).
Here’s my Asterisk configuration files, along with the SPA3102 settings:
1. Asterisk Configuration
sip.conf
[line1]context=outgoinghost=192.168.1.100port=5060username=line1secret=welcometype=frienddtmfmode=rfc2833
[pstn]context=incominghost=192.168.1.100port=5061type=frienddtmfmode=rfc2833insecure=very
[gizmo]context=incominghost=proxy01.sipphone.comusername=YourGizmoUserName or YourGizmoSIPPhoneNumbersecret=YourGizmoPasswordfromuser=YourGizmoUserName or YourGizmoSIPPhoneNumberfromdomain=proxy01.sipphone.comtype=frienddtmfmode=rfc2833insecure=verynat=yes
extensions.conf
[incoming]exten => s,1,NoOP(${CALLERID}) ; show the caller ID info in the consoleexten => s,n,ResponseTimeout(10)exten => s,n,DigitTimeout(5)exten => s,n,SetMusicOnHold(native)exten => s,n,Ringingexten => s,n,Answerexten => s,n,Dial(SIP/line1,15)exten => s,n,Playback(nbdy-avail-to-take-call)exten => s,n,Directory(default,extensions,f)include => extensions
[outgoing]; international/long-distance speed dials - through gizmoexten => 01,1,Dial(SIP/0111111111@gizmo,60)exten => 02,1,Dial(SIP/0112222222@gizmo,60)
; local speed dials - through pstnexten => 11,1,Dial(SIP/1111111111@pstn,30)exten => 12,1,Dial(SIP/2222222222@pstn,30)
; pass throughexten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@pstn,30)exten => _1800XXXXXXX,1,Dial(SIP/${EXTEN}@pstn,30)exten => _1888XXXXXXX,1,Dial(SIP/${EXTEN}@pstn,30)
include => extensions
[extensions]exten => 101,1,VoiceMail(u101@default) ; Mailbox 1exten => 102,1,VoiceMail(u102@default) ; Mailbox 2
exten => 500,1,VoiceMailMainexten => 600,1,Directory(default,extensions,f)exten => 800,1,Background(if-u-know-ext-dial)exten => 800,n,WaitExten(5)exten => 900,1,Playback(demo-moreinfo)exten => 900,n,Hangup
; Invalidexten => i,1,Playback(pbx-invalid)exten => i,n,WaitExten(5)
; Timeoutexten => t,1,Playback(vm-goodbye)exten => t,n,Hangup
2. SPA3102 Settings:
A. Line1 Tab:
Proxy and Registration:
- Display Name: spa3102-line1
- User ID: line1
- Password: welcome
Subscriber Information:
- Display Name: spa3102-line1
- User ID: line1
- Password: welcome
Gateway Accounts:
- Gateway 1:
- GW1 Auth ID: line1
- GW2 Password: welcome
Dialplan:
- Dial Plan: (911S0|1800xxxxxxxS0|011xx.||x.)
B. PSTN Tab:
Proxy and Registration:
- Proxy: IP of Asterisk Server
- Register: no
- Make Call without Reg: yes
- Ans Call without Reg: yes
VoIP-To-PSTN Gateway Setup:
- VoIP-To-PSTN Gateway Enable: yes
- VoIP Caller Auth Method: none
- VoIP Caller Default DP: none
PSTN-To-VoIP Gateway Setup:
- PSTN Ring Thru Line 1: no
- PSTN CID For VoIP CID: yes
FXO Timer Values (sec):
Dialplan: