[rotary500]

After setting up and using Asterisk for 6 months I recently got an SPA 3000. I am also using two SPA 2000 ata's and everything works great, but I have been so confused on what to do to set up an spa 3000. I see numerous posts but no real step-by-step and explanation of what needs to be done. There's a configuration wizard for the box, but nothing for Asterisk. So far I have easily set up the FXS port as an extension, that works great! But what has to be done to get the FXO port to recieve calls, and become a trunk? I see mentions of a dialplan, would that go in extensions.conf? Please if anyone has a description of what needs to be done to get the FXO port to work, or a link to related posts can you let me know? THanks in advance!

[AlexanderBell]

Originally Posted by rotary500

After setting up and using Asterisk for 6 months I recently got an SPA 3000. I am also using two SPA 2000 ata's and everything works great, but I have been so confused on what to do to set up an spa 3000. I see numerous posts but no real step-by-step and explanation of what needs to be done.

I had the same problems you're facing. Here's the step-by-step that worked for me with Asterisk.

[bodosom]

I think step-by-step guides are of little value since they seem to assume a lot about your installation. The key items are:
PSTN-To-VoIP Gateway Enable: yes
PSTN Ring Thru Line 1: no
PSTN Caller Default DP: N
Dial Plan N: (S0<:@asterisk.ip.address>)
PSTN Caller Auth Method: none
Off Hook While Calling VoIP: no

The details depend on your sip.conf file. I was inspired by the Mundy article but since I don't run AAH and haven't done the other bits he describes they are of limited utility.

edit:
I think you have to be running version 3 firmware on the Sipura. I run 3.1.5

[rotary500]

Still No luck. The spa 3000 can tell when the pstn line is ringing or when it's idle, but it doesn't go through Asterisk, I have the asterisk terminal open, and nothing shows up at all for incomming calls. No luck getting out on the fxo port either. Extension 205 (fxs port) still works great. I downloaded another asterisk manual so the adventure continues!~ At least I know what it IS possible to use a pstn line through Asterisk since some have gotten it to work Maybe I will build a computer, install asterisk @ home and follow the page given step by step. Once working I could read the configs and patch mine to make it work. There has to be a better way.

FXS ports seem easy to me, set up the box with an IP, user name and password, create the extensions in Asterisk. The FXO ports just drive me nuts.

[artarzi]

I too have faced the same problem. I solved it by call-forwarding all calls to one of the DID's.. coupled with the

In the PSTN line page set
PSTN CID For VoIP CID: yes
I do not use the Dialplan setting. I use call fowarding

In the PSTN users page set
Cfwd Sel1 Caller: *
and
Cfwd Sel1 Dest: did_number@asterisk-IP-address

I know this should not be required, but it's what I've done.

[bodosom]

Still No luck.

I've found sip debug ip to be quite useful in these sorts of circumstances. By the way my dialplan says <tag@asterisk.host>. You'll be looking for INVITES and want to see a line like:

Looking for <USER> in <CONTEXT> ...

and you'll need the user in your sip.conf file.

You should also see a VoIP call in the PSTN status.

[rotary500]

If I call forward, wouldn't it answer the PSTN immediately? I would prefer to see a caller ID first and then decide whether to answer...

[rotary500]

By the way with the debug, does anyone know what this means?

Sip read:
SIP/2.0 200 OK
To: <sip:*sipura address*:5061>;tag=d55a475b13169b67i1
From: "Unknown" <sip:Unknown@*sipura address*>;tag=as066c7914
Call-ID: 0157be992649caaf13c3afb82293eca4@*asterisk address*
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP *asterisk address*:5060;branch=z9hG4bK21af7d6b
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

I changed the ip's to sipura address and asterisk address for privacy

the next line looks like this
Destroying call '368506cc6ee8f48c732fbf1e5044e6ee@*asterisk address*'

[bodosom]

An answer to your question is that this is probably from a heartbeat option (e.g. qualify=yes).

[bodosom]

Originally Posted by rotary500

I would prefer to see a caller ID first and then decide whether to answer...

Given an appropriate value for PSTN Answer Delay: the dialplan forwarding technique reports CID. Since I don't use selective forwarding I don't know if it works or not.