FreePBX + SPA 3102 Problems

Ok, from my other posts here you will no doubt no I have been playing
FreePBX on a Raspberry PI and now just got myself a Linksys SPA 3102. I've
got it all working, sort off. The 3102 is all logged into FreePBX ok as an
extension and a truck (PSTN Line). I've followed the instructions Graham.
posted but I getting the 'All Circuits Are Busy Now' message when I select
the BT line as a destination, it works ok when I set Sipgate for example so
it's an issue with the 3102 settings I think. I've not tested inbound yet -
no cell phone coverage here for one thing!

Thanks BTW to all the help from group members that have helped me so
far - Thank you all !

Dave

 

Has the PSTN line ('Line 2') registered with the server OK? Does the web
interface show 'registered'?

Here is a fragment of the script I use to provision mine. Ignore the
code, but ensure you have these settings:

call item 'VoIP-To-PSTN-Gateway_Enable_2_','Yes'
call item 'VoIP_Caller_Auth_Method_2_','None'
call item 'VoIP_PIN_Max_Retry_2_',3
call item 'One_Stage_Dialing_2_','Yes'
call item 'Line_1_VoIP_Caller_DP_2_','None'
call item 'VoIP_Caller_Default_DP_2_',2
call item 'Line_1_Fallback_DP_2_','None'
call item 'VoIP_Caller_ID_Pattern_2_',''
call item 'VoIP_Access_List',''
do i = 1 to 8
call item 'VoIP_Caller_'i'_PIN_2_',''
call item 'VoIP_Caller_'i'_DP_2_','None'
end

 

Yes, Both Line 1 Status and PSTN Line Status are showing registered the
other settings are now the same as yours - still have the All Circuits Are
Busy Now (ACABN).

 

Red face time . I was sat on the floor in our hallway and just checking
the leads and thinking then I discovered the Line lead was not plugged all
the way in !!
I knew I didn't want to plug it all the way in just far enough to make
contact. This is because the 'barb' is missing from that line lead and I
didn't want to press it all the way in because it's difficult to remove.

I can now make outbound calls ! Just discovered caller ID is not working
from another extension, small point. Also takes ages to dial from analogue
phone - I think this an issue with dial_timeout in FreePBX

Dave.

 

No inbound either - testing with BT's 17070 ringback test ... there appears
to be loop current flowing looking at the voice info page, so the 3102 is
recognising the ringing but not answering it or passing it on ?

 

Any use?

call item 'PSTN-To-VoIP_Gateway_Enable_2_','Yes'
call item 'PSTN_Caller_Auth_Method_2_','None'
call item 'PSTN_Ring_Thru_Line_1_2_','No'
call item 'PSTN_PIN_Max_Retry_2_',3
call item 'PSTN_CID_For_VoIP_CID_2_','Yes'
call item 'PSTN_CID_Number_Prefix_2_',''
call item 'PSTN_Caller_Default_DP_2_',g.pstn_dp
call item 'Off_Hook_While_Calling_VoIP_2_','No'
call item 'Line_1_Signal_Hook_Flash_To_PSTN_2_','Disabled'
call item 'PSTN_CID_Name_Prefix_2_',''
call item 'PSTN_Caller_ID_Pattern_2_',''
call item 'PSTN_Access_List',''
do i = 1 to 8
call item 'PSTN_Caller_'i'_PIN_2_',''
call item 'PSTN_Caller_'i'_DP_2_','1'
end

 

Yes, here are my settings ..
http://farm6.staticflickr.com/5476/10498580965_0ae1b52586_o.jpg

 

Hi Dave,
Do you still see a long delay if you append # at the end?

As the FXO and FXS are behind a common network interface and therefore
share a MAC they need to have different ports assigned, and by default
the SPA has 5061 assigned to the FSO. In the freepbx trunk, you should
have a line in PEER Details that says

port=5061

Here are my PEER Details in full


disallow=all
allow=ulaw
canreinvite=no
context=from-trunk
dtmfmode=rfc2833
host=dynamic
incominglimit=1
nat=never
port=5061
qualify=yes
secret=????
type=friend
username=1-pstn


SSH into the RPi and issue "asterisk -rvvvvv" at the root prompt,
then see what it spews out when the POTS line rings.

 

That looks fine, and I take it you modified the "Dial Plan 2" further
up the page to (S0<:01234567890>) matching the DID of your inbound
route in Freepbx?

And the inbound route has a valid destination set?

As I said, SSH into the console and ring the line.

 

Hi Graham,

Yes, a # appended speeds it up, I think it can be adjusted with
dial_timeout somewhere but can cause problems.

Peer details exactly like yours, I'll look into a more verbose output .

my lash-up !

http://farm6.staticflickr.com/5529/10498054304_09feec4d57_k.jpg


Dave.

 

Erm no, must have missed that, not sure what to set to ? I have a route
ANY/ANY to pick it up, then sends it to the 3102's FXS port extension.

 

A man after my own heart Dave, most of my projects look as messy as
that, but I'm quite proud of the neatness of my final configuration

http://www.flickr.com/photos/g3zvt/10499539866/lightbox/

Note the second PAP2 has been removed, three extensions are
sufficient. And the disaster recovery spare SD card, very impotent.

I bet you don't work much DX with that PLA going.

 

Not a PLA, won't have them in the house the PI puts out a bit of noise on
lower bands as it is. The one in the right socket is a Maplin power meter
(16Watts Draytek+Rpi+3102).
The one in left socket is the dect phone power supply.

Dave

 

OK I have just tried a few things and with a ANY/ANY route set it does
work, but not when dial plan 2 is set to (xx.) (The caller hears a
DISA dialtone)

If I set it to
(S0<:0>) (any digit(s) can follow the colon), then it does follow the
"/" route.

Make a specific route for it by putting the same number in the Freepbx
inbound route DID, as in the SPA PSTN line Dial plan 2, The obvious
string to use is your POTS phone number so for example
01234567890 in the DID
and (S0<:01234567890>) in DP2

Try that.

 

Apology tendered!
I mistook that power meter for one.

I have a similar power meter, an HO1569 that I picked up for under a
fiver IIRC.

 

Hi Graham,
Just tried that and it seemed to work, sort of, When I dial 17070 from the
line directly then quickly swap the over to the 3102 to receive the call
that works ok, and the 3102 FXS phone rings as directed. However, when I
dial out via the 3102 I get a weird BT message about 'Sorry, if you want to
access your messaging service from home you must dial 1571 - goodbye !'.
Looking at the CDR record it appears to dial 17070 ?. It would appear that
the 3102 is really dialling something else ??

The outbound route is very simple just a dot . to wildcard everything to the
3102 route out, nothing prefixed or appended.

 

Can't you use a mobile to make the calls?

You seem to have discovered a "feature" in the BT system.
The Betamax trunk is presenting the same CLI as the actual line you
are dialling hence the message from 1571.

I hadn't realised it did that, (never had 1571) but I can see why it
might do that.
I think you can dial an 0800 and cancel 1571, It is unnecessary with
Asterisk , you will agree!

So, use a mobile,
or present a different number (or no CLI).


Your getting there!

 

You're not your.
Sorry.

 

Morning Graham,

No mobile coverage at home, Orange or O2, have to up the hill - not that
useful ..

That's what I thought never heard that BT announcement before - I'll try to
record it !, not really sure why it's doing it. I thought 1571 would play
nicely with FreePBX.
It is the DP2 that is causing it because if I change it back to how it was
were back to normal.

I have it set up very simply, any number goes via the 3102 to BT line and
the other way back in to the 3102. I knew the 3102 was a beast to configure
but not realised how much so. I'll try to document my findings to add to the
list of other 3102 setup pages I found - some more useful than others.

My wife keeps saying I have more patience than she does and I will get there
in the end !! - I hope so...