[asterisk-users] Asterisk + VOSP account working configuration?

[Gilles]

On Tue, 14 Dec 2010 11:19:48 -0600, Lyle Giese <lyle at lcrcomputer.net>
wrote:
>You are setting up a SIP trunk from your VOSP provider(whatever VOSP
>is). It dials your phone number. So whatever you dial from your cell
>phone is the extension that this trunk should land at.
>
>'s' is not an extension. It's a placeholder for the steps in your dial plan.

Thanks Lyle, but this document showed a working example using the "s"
extension, and I did get it working using this:

www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx

However, I noticed that order is important: When [vosp_incoming] came
before [vosp_outgoing], I got a BUSY signal when calling the VOSP
number, while I could still make outgoing calls to the POTS:

;================== sip.conf
[ general]
port = 5060
bindaddr = 0.0.0.0

externip=<my public IP>
localnet=192.168.0.0/24

disallow=all
allow=ulaw
allow=alaw
allow=gsm

register => myaccount:mypasswd at myvosp.com

;IMPORTANT: outgoing must be BEFORE incoming
[vosp_outgoing]
type=peer
host=myvosp.com
username=myaccount
secret=mypasswd
fromuser=myaccount
fromdomain=myvosp.com
nat=yes
canreinvite=no

[vosp_incoming]
type=peer
host=myvosp.com
nat=yes
canreinvite=no
context=from_vosp

[6011]
type=friend
secret=my_password
host=dynamic
context=internal
;calls go through Asterisk on same LAN
nat=no

;================== extensions.conf
[general]
static=yes
writeprotect=yes
clearglobalvars=no
autofallthrough=yes

[from_vosp]
exten => s,1,Dial(SIP/6011)
exten => s,n,Hangup()

[to_vosp]
;All numbers starting with 0 are sent to VOSP
exten => _0.,1,Dial(SIP/ippi_outgoing/${EXTEN})
exten => _0.,n,Hangup()

[internal]
exten => 6011,1,Dial(SIP/6011)
exten => 6011,n,Hangup()
exten => 6012,1,Dial(SIP/6012)
exten => 6012,n,Hangup()

include => to_vosp
;================== 

If someone knows of a thorough article/book on how configuration files
work in Asterisk, I'm interested.

Thank you.