One way audio problem with asterisk

I have a call come in to PSTN, this call gets converted as a SIP call and comes to an asterisk switch, asterisk plays back an IVR, the user selects 2, asterisk makes a sip call of 7162275568 back to device that send the inbound SIP call. Yes we are using asterisk as an IVR platform.

The problem is one way audio. Asterisk is dialing the number but the called is not able to hear the caller.

I have included below the extensions.conf and the sip.conf so that you'll can tell me what might be the issue here. Also if you can tell me what should be deleted it would be very helpful.

Extensions.conf

[general]
static=yes
writeprotect=yes
clearglobalvars=no

[telecom]
include => local-extensions

exten => h,1,Hangup
exten => i,1,AbsoluteTimeout(15)
exten => i,2,Congestion
exten => i,3,Hangup
;
[customer]
include => local-extensions

; Record voice file to /tmp directory
exten => 205,1,Wait(2)
exten => 205,2,Record(/tmp/asterisk-recording:wav)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording)
exten => 205,5,wait(2)
exten => 205,6,Hangup

[local-extensions]
include => customer_mainmenu
exten => 1000,1,Goto(customer_mainmenu,s,1)

[customer_mainmenu]
include => local-extensions
exten => s,1,Answer
exten => s,n,Background(customer_intro)
exten => s,n,WaitExten

;Dial said extensions
exten => 2,1,Dial(SIP/1111111111@telecom,30)
exten => 3,1,Dial(SIP/1111111111@telecom,30)
exten => 4,1,Dial(SIP/1111111111@telecom,30)
exten => 5,1,Dial(SIP/1111111111@telecom,30)

[default]
include => customer

sip.conf
[general]
context=telecom ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
allow=ulaw ; Allow codecs in order of preference
allow=alaw ; Allow codecs in order of preference

[telecom]
canreinvite=no
host=xx.y.z.mno
type=peer
user=phone

[customer]
canreinvite=no
context=from-trunk
dtmf=inband
dtmfmode=rfc2833
fromdomain=xx.y.z.mno
host=xx.y.z.mno
insecure=very
type=user
user=phone

--SunMoonStar

Hi I cant see and localhost settings or nat settings.

Is the box behind a nat'd router ?

Ian