hi
i installed asterisk 1.0.9 onder fedora core 3 with Valid Ip and without any ip nating
i want transfer te outgoing calls to one proxy sip server .while i try call to any global pstn telphone number (for example my desk telephone number) it ringing but NO SOUND TRANSFER
my extension.conf configuration file is:
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g2 ; Trunk interface
[test]
exten => 04567,1,Answer
exten => 04567,2,Dial(Sip/2002,10,tr)
exten => 04567,3,Hangup
exten => _X.,1,Dial(SIP/${EXTEN}@64.71.145.237)
exten => t,1,Hangup
exten => T,1,Hangup
exten => 2001,1,Dial(SIP/2001@2001)
exten => 2002,1,Dial(SIP/2002@2002)
and my sip.conf configuratin is :
[general]
nat=yes
canreinvite=no
context=test
autocreatepeer=yes
port=5060
bindport=5060
bindaddr=217.218.46.234
srvlookup=yes
maxexpirey=180
tos=reliability
defaultexpirey=160
allow=all
allow=g729
;allow=gsm
;allow=alaw
allow=ilbc
allow=g723.1
;allow=ulaw
register => user:pass@sip.carriecall.com/04567
[sip.carriecall.com]
type=peer
secret=pass
username=user
host=sip.carriecall.com
user=user
allow=all
allow=g729
allow=ulaw
allow=alaw
fromuser=04567
fromdomain=carriecall.com
nat=yes
dtmfmode=rfc2833
context=test
insecure=very
canreinvite=no
[sip-in]
type=user
context=test
dtmfmode=rfc2833
allow=g729
allow=all
allow=ulaw
allow=ilbc
allow=g723.1
nat=yes
[2001]
type=friend
secret=2001
username=2001
host=dynamic
context=test
allow=ilbc
allow=g729
dtmfmode=rfc2833
canreinvite=no
nat=yes
–END–
i install g723 & g729 codec from kvin.lv/pub/Linux/Asterisk/
and copy this to /module asterisk directory ,my iptables settings is
iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT
iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT
I don 't know how do it ?