betadraco*CLI> 
<-- SIP read from 64.112.124.99:5060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.170.1.25:5060;branch=z9hG4bK7e520a0c;rport=5060;received=209.172.44.113
From: "someone" <sip:5143332222@209.170.1.25>;tag=as550e98e4
To: <sip:13036298344@64.112.124.99>
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines) ---
betadraco*CLI> 
<-- SIP read from 64.112.124.99:5060: 
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 209.170.1.25:5060;received=209.172.44.113;branch=z9hG4bK7e520a0c;rport=5060
From: "someone" <sip:5143332222@209.170.1.25>;tag=as550e98e4
t: <sip:13036298344@64.112.124.99>;tag=as5648a006
i: 0d53ad0842f9d6bc35ca5e624bb2e48d@209.170.1.25
CSeq: 102 INVITE
User-Agent: STC 7.0T
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
m: <sip:6913036298344@75.126.46.154:5065>
c: application/sdp
l: 218

v=0
o=root 22417 22417 IN IP4 75.126.46.154
s=session
c=IN IP4 75.126.46.154
t=0 0
m=audio 15520 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (11 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 75.126.46.154:15520
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    -- SIP/netiqsys-54d5 is making progress passing it to SIP/209.169.1.2-b6707da0
betadraco*CLI> 
<-- SIP read from 64.112.124.99:5060: 
SIP/2.0 200 OK
v: SIP/2.0/UDP 209.170.1.25:5060;received=209.172.44.113;branch=z9hG4bK7e520a0c;rport=5060
Record-Route: <sip:64.112.124.99;lr=on;ftag=as550e98e4;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAc->
From: "someone" <sip:5143332222@209.170.1.25>;tag=as550e98e4
t: <sip:13036298344@64.112.124.99>;tag=as5648a006
i: 0d53ad0842f9d6bc35ca5e624bb2e48d@209.170.1.25
CSeq: 102 INVITE
User-Agent: STC 7.0T
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
m: <sip:6913036298344@75.126.46.154:5065>
c: application/sdp
l: 218

v=0
o=root 22417 22418 IN IP4 75.126.46.154
s=session
c=IN IP4 75.126.46.154
t=0 0
m=audio 15520 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (12 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 75.126.46.154:15520
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:64.112.124.99;lr=on;ftag=as550e98e4;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAc->
set_destination: Parsing <sip:64.112.124.99;lr=on;ftag=as550e98e4;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAc-> for address/port to send to
set_destination: set destination to 64.112.124.99, port 5060
Transmitting (NAT) to 64.112.124.99:5060:
ACK sip:6913036298344@75.126.46.154:5065 SIP/2.0
Via: SIP/2.0/UDP 209.170.1.25:5060;branch=z9hG4bK00e8b56d;rport
Route: <sip:64.112.124.99;lr=on;ftag=as550e98e4;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAc->
From: "someone" <sip:5143332222@209.170.1.25>;tag=as550e98e4
To: <sip:13036298344@64.112.124.99>;tag=as5648a006
Contact: <sip:5143332222@209.170.1.25>
CSeq: 102 ACK
User-Agent: Voicemeup 1.2(vBS.1)
Max-Forwards: 70
Remote-Party-ID: "someone" <sip:5143332222@209.170.1.25>;privacy=off;screen=no
Content-Length: 0


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