<--- SIP read from UDP:192.168.1.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK49e18d8d
To: <sip:elartey@192.168.1.11>;tag=izgis
From: "emma" <sip:emma@192.168.1.155>;tag=as0dff0bb4
Call-ID: 47cd156e10db259324396a862a027de7@192.168.1.155:5060
CSeq: 104 INVITE
Contact: <sip:elartey@192.168.1.11>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.4.2
Supported: replaces,norefersub
Content-Length: 206

v=0
o=twinkle 1923345054 212865941 IN IP4 192.168.1.11
s=-
c=IN IP4 192.168.1.11
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.11:8000
set_destination: Parsing <sip:elartey@192.168.1.11> for address/port to send to
set_destination: set destination to 192.168.1.11:5060
Transmitting (no NAT) to 192.168.1.11:5060:
ACK sip:elartey@192.168.1.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK20df97dc
Max-Forwards: 70
From: "emma" <sip:emma@192.168.1.155>;tag=as0dff0bb4
To: <sip:elartey@192.168.1.11>;tag=izgis
Contact: <sip:emma@192.168.1.155:5060>
Call-ID: 47cd156e10db259324396a862a027de7@192.168.1.155:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.5.0
Content-Length: 0