<--- SIP read from 64.154.41.100:5060 --->
INVITE sip:4018304222@64.150.177.178 SIP/2.0
Max-Forwards: 67
Session-Expires: 3600;refresher=uac
Supported: timer
To: <sip:4018304222@64.154.41.100>
From: <sip:5855555555@64.154.41.100>;tag=3475565976-772087
Contact: <sip:5855555555@64.154.41.100:5060>
Remote-Party-Id: <sip:5855555555@192.168.120.28;user=phone>;party=calling;id-type=subscriber;privacy=off;screen=yes
Call-ID: 569051-3475565976-772055@msx71.mydomain.com
CSeq: 1 INVITE
Via: SIP/2.0/UDP 64.154.41.100:5060;branch=z9hG4bK0cf8e828e1f3c810fcacfb8e4008d25d
Call-Info: <sip:207.2.123.180>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 313

v=0
o=msx71 2147483647 2147483647 IN IP4 64.154.41.100
s=sip call
c=IN IP4 64.154.41.101
t=0 0
m=audio 39510 RTP/AVP 0 18 8 101
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=ptime:20

<------------->
--- (14 headers 14 lines) ---
Sending to 64.154.41.100 : 5060 (no NAT)
Using INVITE request as basis request - 569051-3475565976-772055@msx71.mydomain.com
Found no matching peer or user for '64.154.41.100:5060'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 64.154.41.101:39510
Found audio description format telephone-event for ID 101
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 64.154.41.101:39510
Looking for 4018304222 in inbound (domain 64.150.177.178)
list_route: hop: <sip:5855555555@64.154.41.100:5060>

--- Transmitting (no NAT) to 64.154.41.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.154.41.100:5060;branch=z9hG4bK0cf8e828e1f3c810fcacfb8e4008d25d;received=64.154.41.100
From: <sip:5855555555@64.154.41.100>;tag=3475565976-772087
To: <sip:4018304222@64.154.41.100>
Call-ID: 569051-3475565976-772055@msx71.mydomain.com
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:4018304222@64.150.177.178>
Content-Length: 0


<------------>
Audio is at 64.150.177.178 port 17122
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 64.154.41.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.154.41.100:5060;branch=z9hG4bK0cf8e828e1f3c810fcacfb8e4008d25d;received=64.154.41.100
From: <sip:5855555555@64.154.41.100>;tag=3475565976-772087
To: <sip:4018304222@64.154.41.100>;tag=as3ba96025
Call-ID: 569051-3475565976-772055@msx71.mydomain.com
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:4018304222@64.150.177.178>
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 22864 22864 IN IP4 64.150.177.178
s=session
c=IN IP4 64.150.177.178
t=0 0
m=audio 17122 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
64-150-177-178*CLI>
<--- SIP read from 64.154.41.100:5060 --->
ACK sip:4018304222@64.150.177.178 SIP/2.0
Max-Forwards: 67
To: <sip:4018304222@64.154.41.100>;tag=as3ba96025
From: <sip:5855555555@64.154.41.100>;tag=3475565976-772087
Contact: <sip:5855555555@64.154.41.100:5060>
Call-ID: 569051-3475565976-772055@msx71.mydomain.com
CSeq: 1 ACK
Via: SIP/2.0/UDP 64.154.41.100:5060;branch=z9hG4bK9486ea87c4582bdf1e98ae30c035c60b
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
64-150-177-178*CLI>
<--- SIP read from 64.154.41.100:5060 --->
BYE sip:4018304222@64.150.177.178 SIP/2.0
Max-Forwards: 67
To: <sip:4018304222@64.154.41.100>;tag=as3ba96025
From: <sip:5855555555@64.154.41.100>;tag=3475565976-772087
Contact: <sip:5855555555@64.154.41.100:5060>
Call-ID: 569051-3475565976-772055@msx71.mydomain.com
CSeq: 2 BYE
Via: SIP/2.0/UDP 64.154.41.100:5060;branch=z9hG4bK561ceb03cfabfd8815e55400cd3978dc
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Sending to 64.154.41.100 : 5060 (no NAT)

<--- Transmitting (no NAT) to 64.154.41.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.154.41.100:5060;branch=z9hG4bK561ceb03cfabfd8815e55400cd3978dc;received=64.154.41.100
From: <sip:5855555555@64.154.41.100>;tag=3475565976-772087
To: <sip:4018304222@64.154.41.100>;tag=as3ba96025
Call-ID: 569051-3475565976-772055@msx71.mydomain.com
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

64-150-177-178*CLI>
<------------>
Really destroying SIP dialog '569051-3475565976-772055@msx71.mydomain.com' Method: BYE
Really destroying SIP dialog 'a51e63d1-624ea524-90ee82@216.115.69.131' Method: OPTIONS