[PhoneBoy]

Now available for your testing pleasure:

Voxilla - Linksys SPA 3xxx Configuration Wizard for Asterisk

Might also work with other PBX-type systems as well. It does generate some basic sample configs for Asterisk. Feedback welcome, of course.

[jbd]

Thanks for this great tool. But what does it set up the SPA3K to do?

I was assuming that if I picked up the phone attached Line1 on the SPA3K, that I would get a dialtone. And that if I were to dial an extension, I'd get either Asterisk voicemail or a busy or something.

Asterisk is telling me the following when I pick up the phone and try to dial 30:

-- Registered SIP '30' at 192.168.56.20 port 5060 expires 3600
-- Saved useragent "Sipura/SPA3000-2.0.9(GWi)" for peer 30
-- Registered SIP '40' at 192.168.56.20 port 5061 expires 3600
-- Saved useragent "Sipura/SPA3000-2.0.9(GWi)" for peer 40
Setting NAT on RTP to 0
Stopping retransmission on '45ca34e46d681d964b0b163741351504@192.168.56.8' of Request 102: Found
Auto destroying call 'a6031127-e1997bda@192.168.56.20'
Auto destroying call '8e158c3-800878ee@192.168.56.20'
Sending VNAK

192.168.56.20 is the SPA3K. 192.168.56.8 is the Asterisk system.

Where do I turn on logging on the SPA to find out what's going on?

[PhoneBoy]

You should be able to dial an extension on your Asterisk, yes, but I did not test it with 2.0.9 firmware. I suggest upgrading to the latest release.

[ichilton]

Hi Phoneboy,

Great resource but I think there is a small bug in the example Asterisk configurations it generates - the section in sip.conf for outgoing calls doesn't have any host or port variables so obviously outgoing calls do not work because it doesn't try to connect to the spa3k. Maybe you could add this in.

Thanks!

--ian

[PhoneBoy]

Can you provide an example of where this is messed up? I thought we had this right, but I could be wrong.

[ichilton]

Hi Phoneboy,

Yeah, you have this:

[pstn-spa3k]
type=peer
auth=md5
secret=password
username=asterisk
fromuser=asterisk
dtmfmode=rfc2833
context=home

There is no host or port parameters in there so when you make an outgoing call, it doesn't know where to connect to.

--ian

[n4zmz]

I've been playing with my SPA3K for a few days now. I noticed the same issue as Ian. Adding the missing host/port parameters does help.

For some reason I can't get the PSTN side of the 3k to work properly.

It could be my test setup. I've plugged the FXO port into the FXS port on a cisco 2600 router. I've verified that if I dial 411 on the cisco FXS port that it will call asterisk and give me the directory. If I dial #9411 on the 3k FXS port, I get a busy.

My next testing will be with a real PSTN line. It will just take longer due to cabling issues.

Also the Asterisk example does not support the 10 digit dialing required in more and more cities. Not that it is a big issue to make the changes....

- Dennis

[ichilton]

Hi,

Did you manage to fix that problem Phoneboy?

Thanks

--ian

[achaniotis]

I tried the SPA3000 config wizard for Asterisk, but could still not get SPA-3000 and Asterisks to work together. After successfully running through the wizard, I also update the sip.conf and extensions.conf with the info provided by the wizard. I then add a [99] section to the extensions.conf (99 is the asterisk extension that is entered in the DP8 during the wizard config) which basically dumps the PSTN caller into voicemail for testing.
When I call the PSTN number on FXO, it just rings and nothing ever happens. In the Sipura Info page, it shows that Line 1 and PSTN have NOT successfully registered, so I suspect that is the problem; however, I have not been able to solve it. Any help would be greatly appreciated, as I am not able to figure it out.

As a related question, does [99] (my Asterisk extension in DP8 as entered in wizard) need to be defined in sip.conf or registered in any way, or is it enough to have it defined in extensions.config?

Thank you!

[PhoneBoy]

99 must be defined in sip.conf as well.