Linksys SPA-3102 and ASterisk/FreePbx

General Help
1

Hi Everyone,

Im new to using Asterisk although Ive had the SPA-3102 for a while - using it in a very basic way.

However, although Ive got everything pretty much working, Ive come across a couple of problems which have me totally stumped.

When I call in via my PSTN line, the 3102 forwards the calls to Asterisk, which in turn sends it to extension 100 which duely rings. Perfect. However, if I dont answer the call then after about 6 rings or so, I hear beep-beep-beep and then silence followed by another beep-beep-beep. This continues about 3 tmes. The call seems to have dropped back from Asterisk by then so I think its the 3102 and it sounds like the pstn-to-voip pin request to me. However, authenticaiton is off and the voip call to Asterisk has already been made!

I foudn the following quote on the internet and wondered if this would help

From firmware release notes:
"… When a PSTN caller is automatically routed to a VoIP destination due to
a) hotline w/o authentication, or b)call forwarding, the SPA
will not take the FXO port off-hook until the VoIP destination answers the
call. If the VoIP call leg fails (busy, etc), the PSTN call will not be
picked up by the SPA at all.
The old behavior for this scenario is that the SPA will off-hook the FXO
port first before calling the VoIP destination. To keep the old behavior,
set the new <A&NBSP;&NBSP;HREF=‘PSTN&NBSP;LINE’>PSTN Line paramter to “yes”
(default is “no”)…"

Does the Sipura answer with the PSTN-to-Voip authenticaiton because it hasnt actually answered the call when forwarding to the gateway?

Secondly, I have to make the solution simple if things go wrong, so I dont leave the family without phones if Im not there.

On the outbound, Ive got various dialplans set up on both 3102 and Asterisk to divert to the PSTN line either at the 3102 level or the Asterisk level - this works fine. On the incoming line, if the power is out on the 3102, then it obviously connects to the FXS port and the phone rings - again great. However, if the 3102 is up, but the Asterisk is down, this doesnt seem to be detected and no phones ring.

Is there anyway I can sort this out?

Thanks

TIm

2

This is hard to diagnose because we have no idea what settings you used. Best thing I can suggest is going to

http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx

and following the instructions there. I don’t know if that will solve your second problem, however, as I have no experience with using a “passthru” extension on these things. Whenever we’ve set one up, we’ve used the PSTN port to get the calls into the Asterisk/FreePBX box, and the line port as an extension off the system, but we treat the two as totally separate, which is apparently not what you want to do. However I think those instructions even mention that situation.

If that page doesn’t cover your situation and you finally do figure out how to do what you want to do, please post what you did to solve the problem!

3

Hi

Thanks for taking the time to reply. My system is set up according to

http://www.zultron.com/2008/11/spa3102-and-freepbx-howto/

however, Ive seen the same/similar instructions at the other sites. Ill look at the site you mentioned and compare, but Id welcome any comments on the link above.

Many thanks

Tim