Linksys spa3000+asterisk 1.6

i am trying my best to configure the linksys spa 3000 on my system (asterisk 1.6) to route the calls.
up to now i managed to edit the sip.conf to connect the spa 3000 and send the call to asterisk.
my issue is that i do not know how to route the call to a particular extension in the extensions.conf.
i would appreciate some help.
the call has been send from the spa as mobile@ ans i would like to send it to extension 500.
also i would like to be able to make calls through the linksys spa by pressing 6

This web site should give you some help; When I watched this episode of Systm it was the first time I had ever heard about Asterisk and I was able to get my system up and running by just watching the show and looking at there show notes. In the show notes they have examples of the sip.conf, extensions.conf and voicemail.conf configuration files with just the info in them that is needed to get the system working. The setup they created is exactly what you are trying to do besides they are using a Sipura spa 3000 which is essentially what you have, Sipura was bought out by Linksys which was bought out by Cisco.

Another site worth mentioning is: You can download the entire book for free as a PDF or view it on line. I own both the first edition which covers Asterisk 1.2 and the second edition which covers Asterisk 1.4 and I have pre-ordered the third edition which will cover Asterisk 1.8. They are the only Asterisk books I own and they are great for when this gets more complicated.

One more interesting video with show notes is this episode of Hak5: and the show notes are here: at there wiki.

Good Luck,

thanks a lot for your reply.
i have seen the video from and it is nice.
i do have some gaps though:
-pstn line on spa3000 they did not show which settings they had.
-how to register the phone 1 (fxs port) as an extension of the sip server. in my case it is extension 508
up to now my config is:
exten => 1-gsm,1,Dial(${SIPPHONES},60,tr)


-on line 1 of spa 3000 i do have the username 508 with passwd 508123, on port 5060.
-on pstn line of spa 3000 i do have the display name gsm with username gsm and passwd 999, on port 5061. the dial plan is (S0<:1-gsm@>)

my problem is that i cannot get the call reach asterisk at all!
any ideas?

I have SPA3102 working successfully, should be very similar to SPA3000 since only difference is supposed to be NAT router. Note the SPA3000 is two separate user agent clients (UAC), one for each port (Line/PSTN), which both require sip.conf credentials.

My process was:

  • get Line/PSTN registered (SPA config and sip.conf)
  • Verify/debug registrations (sip show peers, core set debug 9, core set verbose 9, sip set debug on… enable debug logging in asterisk.conf)
  • Config/debug dialplan (extensions.conf)

Hardest part was getting the SPA config mapped to asterisk configs

Below are my basic configs with your info plugged in, red being that pieces that MUST match. Note there is a dial plan configuration on the SPA as well which may trip you up once you have the registration working, suggested config at bottom



SPA3000 Config:
proxy: astip
register: yes
userid: [color=#FF0000]508[/color]
password: [color=#FF0000]linepasswd[/color]

[PSTN Line]
proxy: astip
register: yes
userid: [color=#FF0000]spa3000_pstn[/color]
password: [color=#FF0000]pstnpasswd[/color]

PSTN Answer Delay: 3 (US callerid arrives after 1st ring so this waits until after CID arrives to place VoIP call)
PSTN-VoIP Gateway Enabled: yes
PSTN Caller Auth Method: none
PSTN RIng Thru Line 1: no
PSTN CID for VoIP CID: yes (you do want callerid info, don’t you?)
PSTN Caller Default DP: 8
PSTN CallerID Pattern: ??? (10 digit callerid in US, would not pass any callerid info without this)
Dial Plan 8: (S0<:508>) (when PSTN line receives a call it is answered and a VoIP call is placed to exten 508, "S and the ()s are required, means: immediately place call replacing dialed number with extension 508, hotline/plar functionality)