VoIP Mailing List Archives

[ravichandran.rajagopal...]

Hi,

I am using Asterisk 1.4.13

The call comes in from a Coppercom soft-switch with a private IP of
192.168.104.2. This gets forwarded to a SessionBorderController with public
IP of x.x.x.x,. This then gets to our asterisk server with a public IP of
8.7.192.58 and a private IP of 192.168.5.0. This asterisk server is behind a
Cisco PIX firewall where we have disabled the SIP and UDP fixups. We have
forwarded all the ports 10000 through 20000 and 5060 tcp/udp ports to the
private IP of the asterisk server.

When I dial 4025901000 the call flows through to the asterisk server where
an IVR is played. When the user chooses 6 the asterisk initiates a new
outbound call to 4023399500 using the inbound trunk i_sip_trunk.
When the user chooses 7 the asterisk initiates a new outbound call to
4023399500 using the outbound trunk o_sip_trunk. The difference between the
i_sip_trunk and o_sip_trunk is that the host ip. The i_sip_trunk has the
host ip of the public IP of SessionBorderController whereas the o_sip_trunk
has the host ip as that of the private ip of the asterisk server.

4023399500 is a test number when I am expecting to hear another IVR "Thank
you for calling Ai Software Solutions we apply technology to solutions blah
blah blah".

Lets find out what happens when the user presses 6 there is extended period
of silence.
When the user presses 7 there is a fast busy received upon pressing the
extension.

Any help rendered from your end would be greatly greatly appreciated. I have
been breaking my head to resolve this problem for a long time now.

Thx
Ravi



Sip.conf file
; SIP Configuration example for Asterisk
[general]
context=i_sip_trunk
context=o_sip_trunk
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
localnet=192.168.5.0/255.255.255.0; private IP of the Cisco PIX 506 firewall
externip=8.7.192.58; public IP of the asterisk server
srvlookup=yes
allow=ulaw
allow=alaw

[i_sip_trunk]
type=peer
nat=no
canreinvite=no
host=x.x.x.x; public IP of the SBC
qualify=yes
dtmfmode=rfc2833
context=default

[o_sip_trunk]
type=peer
nat=no
canreinvite=no
host=192.168.5.10; private IP of the asterisk server
qualify=yes
dtmfmode=rfc2833
context=default

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

[default]
include => agnosco
exten => h,1,Hangup
exten => i,1,Congestion
exten => i,2,Hangup

[agnosco]
include => agnosco_mainmenu
include => i_sip_trunk
include => o_sip_trunk

[agnosco_mainmenu]
exten => s,1,Answer
exten => s,n,Background(agnosco_intro)
exten => s,n,WaitExten

;Dial said extensions
exten => 6,1,Dial(SIP/4023399500 at i_sip_trunk,30)
exten => 7,1,Dial(SIP/4023399500 at o_sip_trunk,30)

[i_sip_trunk]
exten => 4025901000,1,Goto(1000,1)
exten => 1000,1,Goto(agnosco_mainmenu,s,1)

[o_sip_trunk]
exten => _NXXNXXXXXX, 1,Dial(SIP/${EXTEN}@o_sip_trunk,30)
exten => _NXXNXXXXXX, 2, congestion()