Problem with audio

Asterisk Asterisk Support
1

Hi All
I am facing one problem ,iam not able to hear voice through SIP (XLITE on both the side)
After the call ring on server or client side when i get connected iam audible to myself only . Whatever i speak from my microphone iam able to hear that only ,not from the other side(It is same as when asterisk is closed Iam able to hear my voice only ).
Is it due to firewall .i have disabled the firewall, but since connectivity is happening i might be wrong. Iam not sure.

My SIP.conf

[1000] ;SERVER canreinvite=no ; force relaying port=5060 srvlookup=yes callerid="Server" username=1000 secret = password type=friend host=dynamic nat=never ; X-Lite is behind a NAT router qualify=yes record_in=Adhoc record_out=Adhoc dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info ;mailbox=1000 ; Mailbox for message waiting indicator context=sip

[2000] ; CLIENT
canreinvite=no ; force relaying
port=5060
srvlookup=yes
callerid="Client"
username=2000
secret = password
type=friend
host=dynamic
nat=never
qualify=yes
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;mailbox=2000 ; Mailbox for message waiting indicator
context=sip

Exten.conf

[code][sip]
exten => 1000,1,Answer
exten => 1000,n,Dial(SIP/1000)
exten => 1000,n,Hangup

exten => 2000,1,Answer
exten => 2000,n,Dial(SIP/2000)
exten => 2000,n,Hangup
[/code]

2

this can help you

canreinvite=no nat=yes qualify=yes allow=all => or something else

3

I tried it ,but still the same issue. When i activate the command SIP DEBUG .on Asterisk CLI i got this message

[code]<— SIP read from 192.168.1.33:5060 —>

<------------->
— (0 headers 0 lines) Nat keepalive —
localhost*CLI>
<— SIP read from 192.168.1.48:5061 —>

<------------->
— (0 headers 0 lines) Nat keepalive —
Reliably Transmitting (NAT) to 192.168.1.48:5061:
OPTIONS sip:1000@192.168.1.48:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.48:5060;branch=z9hG4bK2c56afd5;rport
From: “asterisk” sip:asterisk@192.168.1.48;tag=as192a77be
To: sip:1000@192.168.1.48:5061
Contact: sip:asterisk@192.168.1.48
Call-ID: 5d14acbe330fa2df0803fa1a2107afe9@192.168.1.48
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 16 May 2008 05:41:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

localhost*CLI>
<— SIP read from 192.168.1.48:5061 —>
SIP/2.0 200 Ok
v: SIP/2.0/UDP 192.168.1.48:5060;branch=z9hG4bK2c56afd5;rport
f: “asterisk” sip:asterisk@192.168.1.48;tag=as192a77be
t: sip:1000@192.168.1.48:5061;tag=2070127884
m: sip:1000@192.168.1.48:5061
i: 5d14acbe330fa2df0803fa1a2107afe9@192.168.1.48
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY
CSeq: 102 OPTIONS
Server: X-Lite release 1105d
l: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘5d14acbe330fa2df0803fa1a2107afe9@192.168.1.48’ Method: OPTIONS
localhost*CLI>
<— SIP read from 192.168.1.33:5060 —>

<------------->
— (0 headers 0 lines) Nat keepalive —
localhost*CLI>
<— SIP read from 192.168.1.48:5061 —>

<------------->
— (0 headers 0 lines) Nat keepalive —
localhost*CLI>
<— SIP read from 192.168.1.33:5060 —>

<------------->
— (0 headers 0 lines) Nat keepalive —
Reliably Transmitting (NAT) to 192.168.1.33:5060:
OPTIONS sip:2000@192.168.1.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.48:5060;branch=z9hG4bK380a0ea7;rport
From: “asterisk” sip:asterisk@192.168.1.48;tag=as3ffceb35
To: sip:2000@192.168.1.33:5060
Contact: sip:asterisk@192.168.1.48
Call-ID: 362945967fd7e9364402aedf25ec086d@192.168.1.48
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 16 May 2008 05:41:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

localhost*CLI>
<— SIP read from 192.168.1.33:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.48:5060;branch=z9hG4bK380a0ea7;rport
From: “asterisk” sip:asterisk@192.168.1.48;tag=as3ffceb35
To: sip:2000@192.168.1.33:5060;tag=155511475
Contact: sip:2000@192.168.1.33:5060
Call-ID: 362945967fd7e9364402aedf25ec086d@192.168.1.48
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY
CSeq: 102 OPTIONS
Server: X-Lite release 1105d
Content-Length: 0
[/code]