Topic: Linksys SPA-3102 on SAIL 3.1

[schiehallion]

Hello,
Has anybody managed to get a Linksys SPA-3102 working on SAIL (preferably 3.1) as a PSTN-to-VOIP gateway?  If so, any chance of posting some working settings, please?
Thanks,
Phill

[fpausp]

Hi,

Is this your first time with spa-3102 or have you worked with sail < 3.1 before ?

[schiehallion]

I have been trying to get the SPA-3102 working on 3.1 for a couple of months without much joy and even tried on 2.2 and 2.6 without success too 

[fpausp]

Hi,

I use sail-2.2.4-50 and spa-3102 in austria. I will help you to get this working, I hope I can remember on all details.

You must know that you have to use general-sip http://forums.contribs.org/index.php?topic=46179.0

First of all read this http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter253

[schiehallion]

Great, thanks.  I will build up another SAIL box running the same versions as you I can look at migrating it once it is working on that.

[fpausp]

The Link has changed to this: http://sarkpbx.com/twiki/bin/view/Main/DocChapter253

In which country should the spa-3102 run ?

[schiehallion]

I am in the UK.

[fpausp]

Please send me your Mailaddress to nuntium@gmx.at, I will send you a little howto. If this howto works I will make it public.

[schiehallion]

Massive thanks to fpausp for the documentation and getting me just about there.  I now have the FXS and FXO (I hadn't managed to get FXO previously) registered and can make outbound calls   However, when I make an inbound call, Asterisk gives me these two lines but I can't find where the closing bracket needs to be.

Code: [Select]

[Dec 14 21:12:10] WARNING[7973]: chan_sip.c:2489 get_in_brackets: No closing bracket found in 'sip:<myphonenumber@192.168.1.17'
[Dec 14 21:12:10] NOTICE[7973]: chan_sip.c:15121 handle_request_invite: Call from '01480383109' to extension '<myphonenumber' rejected because extension not found.
Or maybe it is advantage that no one can call me!?!

[Jeppe Fugl]

Is the howto public anywhere?

I don't know how to setup spa3102 fxo manually.

Best Regards,
Jeppe

[fpausp]

> Jeppe Fulg
You can find it on: http://www.netztechnik.at/download/linksys/spa3102/

> SARK devs
Please take a look on the little howto to improve the settings if it is necessary.

Best

[Jeppe Fugl]

Hmm, I cannot get everything to work. Maybe you can help me....

I made the setup with spa3102 (192.168.29.65) and sail 3.1 beta3 (192.168.29.61). I followed the pdf guide to setup the spa3102 box.


Line1:
Works fine both directions.

PSTN Line:
I am able to receive incoming calls (which is redirected to line1 through asterisk), but not make calls on the pstn line.

This is the debug log from asterisk:

Code: [Select]

[Feb 15 21:23:51] WARNING[6122]: chan_sip.c:13638 handle_response: Remote host can't match request NOTIFY to call '654b781401293d56140a898153675a3c@192.168.29.61'. Giving up.
    -- Executing [6612xxxx@internal:1] AGI("SIP/401-00000019", "sarkhpe|OutRoute|normal") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=3600)
    -- Channel will hangup at 2011-02-15 21:23:57 UTC.
    -- AGI Script Executing Application: (Dial) Options: (SIP/6612xxxx@peer872||)
    -- Called 6612xxxx@peer872
    -- [b]SIP/peer872-0000001a is circuit-busy[/b]
  == Everyone is busy/congested at this time (1:0/1/0)
    -- AGI Script Executing Application: (Playback) Options: (beep)
    -- <SIP/401-00000019> Playing 'beep' (language 'en-gb')
    -- AGI Script Executing Application: (Playtones) Options: (congestion)
    -- AGI Script Executing Application: (Congestion) Options: ((null))
  == Spawn extension (internal, 6612xxxx, 1) exited non-zero on 'SIP/401-00000019'
    -- Executing [h@internal:1] Hangup("SIP/401-00000019", "") in new stack
  == Spawn extension (internal, h, 1) exited non-zero on 'SIP/401-00000019'
I guess " is circuit-busy" is my problem, but I don't know what to do.

Thoughts about registration:
It seems that asterisk makes the registration to the spa3102 box, when creating the trunk with default parameters. But spa3102 is also trying to connection to asterisk which gives the following error -

[Feb 15 21:09:46] NOTICE[6122]: chan_sip.c:16379 handle_request_register: Registration from '6220xxxx <sip:6220xxxx@192.168.29.61>' failed for '192.168.29.65' - No matching peer found

But in the sail setup everything is fine with latency and on-line check mark.


Any ideas, please 
Best Regards,
Jeppe

[SARK devs]

The SPA wants to register with Asterisk so you should set host=dynamic in the SIP peer entry.   This effectively tells Asterisk to expect a registration.  If you have a registration string in your trunk definition then you should remove it.

Kind Regards

S

[Jeppe Fugl]

Hmm, still the same. This is my peer tab:

Code: [Select]

type=peer
host=dynamic
qualify=3000
canreinvite=no
username=6220xxxx
fromuser=6220xxxx
secret=********
insecure=port,invite
disallow=all
allow=alaw
allow=ulaw
All other tabs are prefilled when creating the trunk. Nothing changes

When connecting the spa3102 i can see this in the asterisk debug console (peer872 is the trunk for 6220xxxx):

Code: [Select]

[Feb 16 12:47:46] NOTICE[6122]: chan_sip.c:16379 handle_request_register: Registration from '6220xxxx <sip:6220xxxx@192.168.29.61>' failed for '192.168.29.65' - No matching peer found
[Feb 16 12:47:46] NOTICE[6122]: chan_sip.c:16379 handle_request_register: Registration from '6220xxxx <sip:6220xxxx@192.168.29.61>' failed for '192.168.29.65' - No matching peer found
[Feb 16 12:47:56] NOTICE[6122]: chan_sip.c:13387 handle_response_peerpoke: Peer 'peer872' is now Reachable. (5ms / 3000ms)
Trunk is offline in the sail page and registration state in the spa3102 web interface says failed.

[SARK devs]

This might help

http://www.sailpbx.com/mediawiki/index.php/SPA3k

Kind Regards

S