<--- SIP read from UDP:192.168.15.117:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.116:5060;branch=z9hG4bK1659f1b7;received=192.168.15.116;rport=5060
From: "cb4833" <sip:asterisk@192.168.15.116>;tag=as55d9b765
To: <sip:469@192.168.15.117>;tag=as6d281c05
Call-ID: 0ef4d1db467e9c2145ff3f422b2fd69b@192.168.15.116:5060
CSeq: 103 INVITE
Server: Digium Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:469@192.168.15.117:5060>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 698410397 698410398 IN IP4 192.168.15.117
s=Digium Gateway
c=IN IP4 192.168.15.117
t=0 0
m=audio 10000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.15.117:10000
list_route: hop: <sip:469@192.168.15.117:5060>
set_destination: Parsing <sip:469@192.168.15.117:5060> for address/port to send to
set_destination: set destination to 192.168.15.117:5060
Transmitting (NAT) to 192.168.15.117:5060:
ACK sip:469@192.168.15.117:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.116:5060;branch=z9hG4bK528a4a09;rport
Max-Forwards: 70
From: "cb4833" <sip:asterisk@192.168.15.116>;tag=as55d9b765
To: <sip:469@192.168.15.117>;tag=as6d281c05
Contact: <sip:asterisk@192.168.15.116:5060>
Call-ID: 0ef4d1db467e9c2145ff3f422b2fd69b@192.168.15.116:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0beta3(11.2.1)
Content-Length: 0


---
    -- SIP/G100-00000085 answered SIP/cb4833-00000084
