[PhoneBoy]

1. This is how one of my SPA-3000s is configured, so yes.

2. Yes, you should, in theory, be able to use both Line 1 and the PSTN simulatenously. They do differ in the SIP port they use (or should).

[bernart]

Got it to work! Turns out that I had the NAT and STUN in the SPA3k enabled and both SPA and * are behind the router. so turned off the NAT and STUN and it worked. The extension in the SIP.conf already had NAT=0

[bernart]

Can * place a call thru line 1?

a. defined an extension in SIP.conf

[line1]
type=peer
host=192.168.0.175 ; spa3k address
port=5062 ; line 1 port
secret=paswordforprovider
username=asknownbyprovider
callerid=" from asterisk"
nat=0


b. in the extension.conf

[global]
voip1=192.168.0.175:5062

exten => _98.,1,Dial(SIP/${EXTEN:2}@line1,60)
exten => _98.,2,Congestion

exten => _99.,1,Dial(SIP/${EXTEN:2}@${VOIP1},60)
exten => _99.,2,Congestion


c. dial plan on line 1 tab (xx.)


Outcome

both calls resulted in the following status message:

"got SIP response 404 not found back from 192.168.0.175
"SIP/192.168.0.175:5062-3a1d is circuit busy"

So the questions:

1. Should be able to call out on the service registered on Line 1?
2. Reasons for the particular status messages


thnx

[PhoneBoy]

Asterisk can ring Line 1 as an extension, but not "make a call through it." You would make a call through the PSTN Line (which has a different SIP port) if you're trying to reach your PSTN line.

[bernart]

thanks, I was trying to call out thru the line 1 voip to avoid dealing with the NAT issues. No getting around that bad boy

[jeremy]

Can you clarify where the SIP credentials are placed in the 3K config? I see a few different places to put this type of info, some of which are not under the PSTN tab. I am trying to get the box to interact with Asterisk.

Thanks,
Jeremy

Originally Posted by PhoneBoy

.
.
.
PSTN Tab:

SIP Credentials (Proxy, User ID, Password) point to an extension in sip.conf. This allows the SPA3000 to make a call to the * box as well as allow the * box to call the SPA-3000.
.
.
.

[bernart]

They are all on the "PSTN Line" tab under "Subscriber Information" and "Proxy and Registration" sections

[syscon]

Originally Posted by PhoneBoy

PSTN Tab:

SIP Credentials (Proxy, User ID, Password) point to an extension in sip.conf. This allows the SPA3000 to make a call to the * box as well as allow the * box to call the SPA-3000.

Can someone explain where and how to enter SIP Credentials?
User ID, Password is obvioius though I'm confused what to enter in Proxy.

I'm trying to configure the SPA3000 so I can make calls via VOIP, PSTN and receive calls. I was able to setup SPA-3000 for FWD but I lost my PSTN dial tone.

There are various instructions on the board on how to setup the unit but none of them are complete and/or explained correctly.

Can someone modify the instruction from:
http://voxilla.com/voxstory39.html
to be applicable to SPA3000 unit?

#Joseph

[syscon]

Originally Posted by muppetmaster

Asterisk Configuration for Outgoing

- extensions.conf

[globals]
PSTN_GW=192.168.1.51:5062

[pstn]
exten => s,1,Dial(SIP/${EXTEN}@${PSTN_GW})

If behind a firewall this should be okay. If not, then of course you would want to configure with HTTP digest in mind.

Can you explain in more detail how to make outgoing calls from * to PSTN Line.
I can not make it to work.
How do I set it up so when I dial "9" it give me PSTN dial tone.

My gatway in global is define correctly
PSTN_GW=10.0.0.154:5062

#Joseph

[PhoneBoy]

As far as the SPA3000 talking to an Asterisk server is concerned, the "proxy" is the Asterisk server's IP address (and port if different from port 5060).

We've talked about doing a configuration guide for the SPA3000 and Asterisk. It's something on the "to do" list.