This is a tough question to format clearly to be understandable..

Im looking for detailed directions how to setup Asterisk with a SIP
softphone on my PC.
Using several SIP accounts for incoming calls, to be forwarded to a mobile number.

This sounds really impossible, however...

I have all information needed for a experienced Asterisk user to
possibly find an answer.

OK first the Asterisk version Im using is Windows based, so it will run on XP.
this can be found here:
http://www.asteriskwin32.com

The SIP device Im using is X-PRO
this can be downloaded here:
http://www.voipuser.org/forum_topic_1987.html

A free SIP account is found here:
http://account.freeworlddialup.com/index_new.php?section_id=94

Once a free FWD account is setup, a free landline number can found here:
http://phone.ipkall.com/ipphone/

###################
next...

I need to setup Asterisk to my FWD account (SIP Proxy) details???

the last and most important answer Im willing to pay for is:

I need Asterisk to be setup using a SIP proxy account and forward all
incoming calls to this specific proxy and number??
(This SIP proxy is my account and is able to make outgoing calls to
any number worlwide)
example of making a outbound call using this account on X-Pro...
account is on SIP proxy 2:
dial:  #20013605154443

This is alot of information for the basic setup of a SIP device, my
main goal is for Asterisk
to be used for either a menu system, which handles all incoming calls
from various SIP accounts and
prompts callers with a menu system, and then redirects callers to a
number which I can change so Im
not always at the computer. Please enquire for anything that may be unclear.

As per my understanding, you are trying to configure asterik PBX
system for following:

1. It should accept calls from different SIP accounts.
2. Callers should be presented a menu.
3. The callers then dial some number.
4. Asterik should collect the digits of the number
3. Then asterik should present that number to a particular FWD SIP account.

To configure multiple SIP accounts for incoming calls, you have to
make 'register' entries for each
SIP account in \cygroot\asterisk\etc\sip.conf. 

The syntax of 'register' directive is given below.

register => user[:secret[:authuser]]@host[:port][/extension]

Using 'register' directive in sip.conf would make asterik software to
connect to different SIP proxies for respective SIP accounts at the
startup. To verify whether the asterik is connected for SIP user
accounts with respective proxies, use the following command on asterik
CLI:

*CLI>  sip show registry

Next step is to write file \cygroot\asterisk\etc\extension.conf so
that callers are presented a menu. I am assuming that the callers
would be first greeted with a welcome message and then they are
prompted to dial a number. This dialed number would then be presented
to FWD SIP account which is capable of making
calls anywhere in world.

(You also need to configure account FWD SIP in sip.conf so that
asterik is able to connect to FWD SIP proxy).

The following section should be added in extension.conf so as to
answer incoming calls, for welcome menu, and then calling FWD account.


[default]
exten => s,1,Ringing
exten => s,2,Wait,2     		; Make caller comfortable with 2 seconds of ringback
exten => s,3,Answer			; Answer the call
exten => s,4,Playback(welcome-msg)	; Play a welcome message
exten => s,5,DigitTimeout,5		; Set Digit Timeout to 5 seconds
exten => s,6,ResponseTimeout,10		; Set Response Timeout to 10 seconds
exten => _.,1,_.,1,Dial(SIP/${EXTEN}@fwd_proxy,,r) ;Call FWD account
using collected digits in EXTEN
exten => h,1,Hangup
exten => t,1,Hangup


This is a sample config which I tried in a simulation environment.

Please ask if some other clarification is needed.

Hello vakulgarg-ga
I will give it try this coming week, as Im getting a better
understanding of these devices and software's.

Hi Vakulgarg-ga

I tried your setup, I was not able to connect to my SIP accounts, this
was found by using CLI & 'sip show registry'

This is what I have so far

SIP: 

[general] 
context=default 
recordhistory=yes 
port=5060 
bindaddr=0.0.0.0 
srvlookup=yes 
externip = 24.86.00.000 

register => 681041:????@fwd.pulver.com 
register => <mandon>:<????>@sip.voipuser.org 

[voipuser] 
disallow=all 
allow=ulaw 
username=<mandon> 
fromuser=<mandon> 
secret=<????> 
type=friend 
context=default 
host=voipuser.org 
fromuser=0844986???? 
fromuser=0844986???? 
fromdomain=voipuser.org 
insecure=very 
qualify=no 
canreinvite=no 


############## 

ext: 

[general] 
static=yes 
writeprotect=no 
exten => <681041>,1,Dial(SIP/voipuser,20) 
exten => <681041>,2,Hangup 

############## 


IAX: 

register => <681041>:<????>@iax.fwdnet.net 
allow=ulaw 

[iaxfwd] 
type=user 
username=<681041> 
context=default 
auth=rsa 
inkeys=freeworlddialup 


############## ############## 

additionally
I will repost the question because of expiry, and amount increase to $20.
The question will need to be broken into parts...

(I now have got the SIP details for all my accounts)

   - all incoming callers have greeting (sound file I will create)
with instruction of menu options / dial 1 for orders, dial 2 for
support etc..

   - the callers for menu option #1 would then be directed to a SIP account:
   (SIP/voipuser,20) Im referencing this instruction parameter from above

   - the callers for menu option #2 would be directed to another SIP

   - after certain amount of rings (15) goto voicemail.

please reply a comment if this is a project for you to complete, Im
hoping to have found solution by end next week

Hi rayzuntheterrible

Are you trying to setup behind a NAT or firewall?

Can you use ethereal (can be downloaded freely from www.ethereal.com)
to see if the SIP 'Register' UDP messages are going
out of your machine on which you have installed asterik to fwd.pulver.com and 
also you are getting the replies back?

This is the first step in the registration with SIP proxy (in ur case FWD).

Please check this.

Vakul

Hi Vakul

Im not using any other hardware or software other than
   XP SP2
&
   XP firewall

I will give the program " ethereal " a try and post back my result