Skip2PBX is 100% compatible with Asterisk. The following configuration has been tested with asterisk 1.4.

1. Define a new SIP trunk in /etc/asterisk/sip.conf

...
register=>asterisk: This e-mail address is being protected from spambots. You need JavaScript enabled to view it .1.1 #Change this with your Skip2PBX IP
...

[skip2pbx]
username=asterisk
type=friend
secret=asterisk
qualify=no
insecure=very
host=192.168.1.1 #Change this with your Skip2PBX IP
fromuser=asterisk
context=from-skip2pbx
canreinvite=no
call-limit=50
...

2. Configure your Skip2PBX to accept trunking from your asterisk

Open your Skip2PBX web interface and go in Channels > SIP > SIP trunks and  create a new trunk with following specs:

Authentication mode	Based on IP (trusted)
Link mode	PBX
Username	asterisk
Remote Side IP	your asterisk IP
Remote side port	5060
DTMF Mode	RFC2833

3. Configure your asterisk extensions.conf depending on your preferences

Example:

...
[from-internal]
exten=>_X.,1,Dial(SIP/${EXTEN}@skip2pbx,30,r)

[from-skip2pbx]
exten=>_!,1,Dial(ZAP/g1,30,r)
...

4. Configure your Skip2PBX routes

Open your Skip2PBX web interface and go in Call routes > Inbound calls or Call routes > Outbound calls and define routes depending on your preferences.

Default outbound route works for 99% of configurations.

PLEASE NOTE: For incoming calls you may need to specify an extension number to route call to.