[Asterisk-Users] forward incoming analog call to SIP?

[Dave Williams]

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) 
answers an incoming call and forwards that call to a SIP softphone (X-lite.)

 Seems all is built/installed okay:

# ztcfg -vv
Zaptel Configuration
======================
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.

I'm pretty new at this and the extensions.conf file is eating my lunch. 
Here are my various config files - maybe someone will take pitty on me 
and point me in the right direction. Needless to say, Asterisk pukes on 
my dialplan when I try and startup. .

(zapata.conf)
context=analog
signalling=fxs_ks
language=en
channel => 1

(sip.conf)
[sip_proxy]
For incoming calls only. Example: FWD (Free World Dialup)
type=user
context=sip

[xlite1]
"Transmit Silence"=YES
type=friend
regexten=1234                 ; When they register, create extension 1234
username=xlite1
callerid="Jane Smith" <5678>
host=dynamic
allow=ulaw
allow=alaw


(extensions.conf)
[general]
static=yes
writeprotect=no

[analog]
include=>test
include=>local

[sip]
include=>test
include=>local

[test]
611,1,echo_test

[local]
exten => 1237,1,Dial(SIP/xlite1,10,t)