asterisk users - Jan 2005 - problems with astcc

hello *'s,
Astcc not workin what is correct format for defining
1-database
2-brands
3-trunks
4-routes
i define all these things but not workin may be i define in wrong 
format.I have FXO card installed.can anyone implement  it and also my 
sip phone generates very loud noise wat is that  i tried several 
settings but not hear any voice just noise.

sip.conf
[general]
context=from-sip
port=5060
bindaddr=192.168.10.186
disallow=al
allow=gsm
musicclass=default
relaxdtmf=yes

register => sipuser:secret@iptel.org

[101]
username=101
type=friend
secret=secret
host=dynamic
defaultip=192.168.10.176 ;ip addr of sip phone
disallow=all
allow=gsm
nat=yes
qualify =1000

[adnan.007]
username=adnan.007
type=peer
secret=secret
host=iptel.org
disallow=all
allow=gsm
nat=yes
qualify=1000

extensions.conf
[general]
static=yes
writeprotect=no
#include => /var/lib/astcc/astcc-exten.conf

[incoming]
exten => _N.,1,Dial(Zap/1)
exten => _N.,2,DeadAGI(astcc.agi)
exten => _N.,3,Hangup

[from-sip]
adnan.007=101
exten => _N.,1,DeadAGI(astcc.agi,${ACOUNTCODE},${EXTEN})
exten => _N.,2,Hangup

 Thanks In Advance.
Adnan Ahmed.

> exten => _N.,1,DeadAGI(astcc.agi,${ACOUNTCODE},${EXTEN}) 
What is the right spelling for Account Code? Why don't you do your home
work first.

Also There is a field missing in the admin.cgi CREATE this field in cdrs
table.

add: callstart CHAR(24) to the cdrs table

There is a patch to fix the cgi at

http://bugs.digium.com/bug_view_page.php?bug_id=0002796 

Seshu

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Adnan
Ahmed
Sent: Thursday, January 13, 2005 3:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] problems with astcc

hello *'s,
Astcc not workin what is correct format for defining 1-database 2-brands
3-trunks 4-routes i define all these things but not workin may be i
define in wrong format.I have FXO card installed.can anyone implement
it and also my sip phone generates very loud noise wat is that  i tried
several settings but not hear any voice just noise.

sip.conf
[general]
context=from-sip
port=5060
bindaddr=192.168.10.186
disallow=al
allow=gsm
musicclass=default
relaxdtmf=yes

register => sipuser:secret@iptel.org

[101]
username=101
type=friend
secret=secret
host=dynamic
defaultip=192.168.10.176 ;ip addr of sip phone disallow=all allow=gsm
nat=yes qualify =1000

[adnan.007]
username=adnan.007
type=peer
secret=secret
host=iptel.org
disallow=all
allow=gsm
nat=yes
qualify=1000

extensions.conf
[general]
static=yes
writeprotect=no
#include => /var/lib/astcc/astcc-exten.conf

[incoming]
exten => _N.,1,Dial(Zap/1)
exten => _N.,2,DeadAGI(astcc.agi)
exten => _N.,3,Hangup

[from-sip]
adnan.007=101
exten => _N.,1,DeadAGI(astcc.agi,${ACOUNTCODE},${EXTEN})
exten => _N.,2,Hangup

 Thanks In Advance.
Adnan Ahmed.
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