[asterisk-users] What really happens between Asterisk and an SPA-3000?

[Larry Alkoff]

I'm trying to get a clear understanding just how calls are routed in a 
mixed SPA3k and Asterisk system.

This is my present (incomplete) understand and I'd appreciate any 
corrections.  I'm especially interested in what happens between Asterisk 
and an SPA3k.

Note:
-----
POTSaudio refers to POTS line audio signal.
SIPaudio refers to sip IP packets.
User is us with the Sipura and Asterisk goodies.
Caller is the outside person calling us.

No power to SPA3k:
------------------
failover mode.
FXO line (CO line) audio is connected directly to FXS line (POTS phones) 
by a relay.

Incoming SIP call to Asterisk:
------------------------------
Handled by Asterisk as SIP to SIP. Call never touches SPA3k.

Incoming POTS call to SPA3k:
----------------------------
POTSaudio converted to SIPaudio signal in SPA3k.
SIPaudio forwarded to PSTN-IN extension on Asterisk server (ext 201 for me).
Asterisk should ring both SIP phones and POTS phones
using context in extensions.conf so User can pickup either.

If POTS phone picked up:
------------------------
Does POTSaudio go directly back and forth over POTS line
or is there a SIP conversion anywhere?

If SIP phone picked up:
-----------------------
SIPaudio from SPA3k Caller is heard by User
SIPaudio from User goes out over PSTN-OUT to SPA3k
which converts SIPaudio to POTSaudio and out Line to Caller.

Outgoing calls:
---------------

If 7 digit local or 911, outgoing context in extensions.conf
routes call to SPA3k and out PSTN-SPA3k gateway.
SIPaudio to SPA3k which converts it to POTSaudio.

Other calls are routed either to SIP extensions
or SIP provider. SPA3k is out of the picture.

Larry
-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux