GOautodial Open Source Omni-channel Contact Center Suite

Hello,
I created a new carrier but when I use it we have no voice.
The carrier made a SIP Trace and he told me that is a NAT issue.
Here the comment :

_The issue is that client sends invite SDP message (which describes RTP connection parameters)
with internal IP. So M2 sends RTP packets to this internal IP and of course these packets do not
reach client, so RTP timeout timer is activated on client system and connection drops.

So solution for this is to inform client to correctly configure system, so that external IP would be

used in SDP description.

INVITE sip:0134410824@136.243.91.205;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK38003487;rport
From: "V9171223490000253536" <sip:asterisk@192.168.1.200>;tag=as5e58234c
To: <sip:0134410824@136.243.91.205;cpd=on>
Contact: <sip:asterisk@192.168.1.200>
Call-ID: 36f4461a261a7b97067da9e76eab995d@192.168.1.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V9171223490000253536" <sip:asterisk@192.168.1.200>;privacy=off;screen=no
Date: Thu, 17 Sep 2015 16:24:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 24037 24037 IN IP4 192.168.1.200
s=session
c=IN IP4 192.168.1.200
t=0 0
m=audio 17338 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Somobody help me ? Where I can modify the internal IP to external IP ?