A forum user who was having trouble emailed me asking for info on my config. I thought I'd post it here for the benefit of anyone else who may be trying to get started. It's probably ok for the SPA3000 as well.
I stress that this is just a rudimentary setup I used to get things talking to one another. You may need to make minor chages to suit the requirements of your local phone system with regard to dial plans. I'm in the US in the Atlanta area where all local calls require 10 digits.
I should also say that I'm a total noob at this stuff. If there are any glaring mistakes or simpler ways I could do things then please feel free to comment.
I'm just going to list the things I changed or may have changed. I've done and undone so many settings that I'm not 100% sure about some things, but if something isn't listed then I'm pretty certain I didn't change it. I didn't configure for any registration in order to keep things simple. Usernames and passwords may not be needed in this state either, but I have them entered so I'm showing that.
One thing I did add to the mix which isn't really necessary is an Xlite softphone config. It's handy for testing purposes. If your don't want to use it you can omit the Xlite stuff from sip.conf and the Internal context in extensions.conf.
I also recommend using Skype with it's free (for now) Skypeout service to test incoming PSTN calls if you don't have another way to call yourself (or a helpful friend to call you). One caveat with Skype: the caller ID info was unreliable--sometimes I'd get it and other times not. For this simple setup that's not an issue, but if you add processing based on caller ID you'll either need to use something else or fake it by stuffing the caller ID variables yourself.
**************** Linksys web interface configuration ****************
*** Line1 Tab ->
Proxy and Registration ->
Proxy: 192.168.0.2
* Put the IP of the PC which runs Asterisk here.
* Not sure about the registration defaults, but mine is set to not register and to allow making and receiving calls without registration.
Subscriber Information ->
Display Name: Line1
Username: line1
Password: <my password>
Dial Plan: (*xx|[3469]11|0|00|[2-9]xxxxxxxxxS0|1xxx[2-9]xxxxxxS0)
* The above dialplan was changed for my area's required 10 digit dialing.
*** PSTN Tab ->
Proxy and Registration ->
Proxy: 192.168.0.2
* Put the IP of the PC which runs Asterisk here.
* Outbound Proxy is left blank. Not sure if this is correct, but for testing purposes it's working.
* Again, not registering and allowing calls in and out without registration.
Subscriber Information ->
Display Name: PSTN
Username: pstn
Password: <my password>
Dial Plans ->
Dial Plan 8: S0<:123@192.168.0.2>
* Found
here.
* This sends all incoming calls to extension 123. Change
192.168.0.2 to match your Asterisk box's IP.
Dial Plan 1: (xx.)
* This is the default and I point some stuff to this dial plan as per below.
VoIP-To-PSTN Gateway Setup ->
VoIP-To-PSTN Gateway Enable: Yes
Line 1 VoIP Caller DP: 1
VoIP Caller Default DP: 1
PSTN-To-VoIP Gateway Setup ->
PSTN-To-VoIP Gateway Enable: Yes
PSTN Ring Thru Line 1: No
* If this is set to yes then someone may hear the ring pickup the phone just as Asterisk taking it. If that happens and Asterisk then tries to ring the FXS port on the 3102 it will be in use, resulting in congestion.
PSTN CID For VoIP CID: Yes
* So I get the caller's CID instead of the SPA3102's.
FXO Timer Values (sec) ->
PSTN Answer Delay: 5
* You may need something longer above. 5 allows enough time on my system to get the caller ID info.
**************** end Linksys web interface stuff ****************
Now on the Asterisk side of things.
;
**************** my sip.conf additions ****************
;; Xlite softphone used for testing.
;; If you don't want to use the Xlite then omit this.
[xlite]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
context=internal
secret=<some_password>
callerid="Your Name" <123-555-9876>
host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
;; Asterisk will ruote calls out to the phone or phones
;; connected to the FXS (Line1) port.
[line1]
type=friend ; so I can dial out from or direct calls to here
host=192.168.0.1
context=line1
username=line1
secret=<some_password>
;mailbox=1000
dtmfmode=rfc2833
disallow=all
allow=ulaw
;; Asterisk will route outgoing calls to this.
[pstn]
; If you're using Asterisk, this section goes into the Outgoing Settings
; for your trunk.
type=friend
;auth=md5
host=192.168.0.1
port=5061
secret=<some_password>
dtmfmode=rfc2833
context=pstn
insecure=very
;
**************** end of my sip.conf additions ****************
;
**************** my entire extensions.conf ****************
[globals]
[general]
[line1]
;; my area uses 10 digit dialing, hence the 10 X's
;; change the following line to suit your needs
exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
;; toll free numbers
exten => _1800XXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
exten => _1888XXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
exten => _1866XXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
[pstn]
;; extension 123 is for incoming PSTN calls
;; in the SPA3102 PSTN tab dialplan S0<:123@192.168.0.2>
;; causes the incoming PSTN to VoIP calls to be redirected
;; to this extension
exten => 123,1,NoOP(${CALLERID}) ; show the caller ID info in the console
exten => 123,n,Ringing()
exten => 123,n,Answer()
exten => 123,n,Playback(silence/1)
exten => 123,n,Playback(pls-wait-connect-call)
exten => 123,n,Wait(3)
;; attempt to connect the call to the FXS (Line1) port
exten => 123,n,Dial(SIP/line1,60)
exten => 123,n,Congestion
[internal]
;; used for Xlite softphone
;; you may omit this context if not using an Xlite softphone
exten => 100,1,Dial(SIP/line1) ; send these call to the FXS (Line1) port
exten => 200,1,Dial(SIP/xlite) ; send these calls to the Xlite softphone
;; my area uses 10 digit dialing, hence the 10 X's
;; change the following line to suit your needs
exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
;; toll free numbers
exten => _1800XXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
exten => _1888XXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
exten => _1866XXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
;
**************** end of extensions.conf ****************
Comments / suggestions appreciated.
Starter SPA3102-Asterisk Setup
Linear Mode
