
<--- SIP read from UDP:192.168.88.102:50445 --->
INVITE sip:8155054487@192.168.20.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK428fa630
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
Date: Tue, 19 Oct 2010 07:59:28 GMT
CSeq: 103 INVITE
User-Agent: CSCO/7
Contact: <sip:2225@192.168.88.102:5060>
Content-Type: application/sdp
Content-Length: 244
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="1a4c3c0c16d4f6e70264137c69ef24bd",nonce="651ddeff",algorithm=MD5

v=0
o=Cisco-SIPUA 12703 10251 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 16756 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (12 headers 11 lines) ---
Sending to 192.168.88.102 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK428fa630;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
CSeq: 103 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8155054487@192.168.20.100>
Content-Length: 0


<------------>
Audio is at 192.168.20.100 port 14816
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK428fa630;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
CSeq: 103 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8155054487@192.168.20.100>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 525914847 525914848 IN IP4 192.168.20.100
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.20.100
t=0 0
m=audio 14816 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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

<------------>

<--- SIP read from UDP:192.168.88.102:50445 --->
ACK sip:8155054487@192.168.20.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK0f81959e
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
Date: Tue, 19 Oct 2010 07:59:29 GMT
CSeq: 103 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="1a4c3c0c16d4f6e70264137c69ef24bd",nonce="651ddeff",algorithm=MD5
Content-Length: 0


<--- SIP read from UDP:192.168.88.102:50445 --->
INVITE sip:8155054487@192.168.20.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK35768fe3
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
Date: Tue, 19 Oct 2010 07:59:35 GMT
CSeq: 104 INVITE
User-Agent: CSCO/7
Contact: <sip:2225@192.168.88.102:5060>
Content-Type: application/sdp
Content-Length: 251
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="1a4c3c0c16d4f6e70264137c69ef24bd",nonce="651ddeff",algorithm=MD5

v=0
o=Cisco-SIPUA 16545 14849 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 192.168.88.102
t=0 0
m=audio 16756 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (12 headers 11 lines) ---
Sending to 192.168.88.102 : 5060 (no NAT)
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.88.102:16756

<--- Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK35768fe3;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
CSeq: 104 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8155054487@192.168.20.100>
Content-Length: 0


<------------>
Audio is at 192.168.20.100 port 14816
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK35768fe3;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
CSeq: 104 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8155054487@192.168.20.100>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 525914847 525914849 IN IP4 192.168.20.100
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.20.100
t=0 0
m=audio 14816 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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

<------------>

<--- SIP read from UDP:192.168.88.102:50445 --->
ACK sip:8155054487@192.168.20.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK5c404c87
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
Date: Tue, 19 Oct 2010 07:59:36 GMT
CSeq: 104 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="1a4c3c0c16d4f6e70264137c69ef24bd",nonce="651ddeff",algorithm=MD5
Content-Length: 0


<--- SIP read from UDP:192.168.88.102:50445 --->
INVITE sip:8155054487@192.168.20.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK1960bfd0
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
Date: Tue, 19 Oct 2010 07:59:37 GMT
CSeq: 105 INVITE
User-Agent: CSCO/7
Contact: <sip:2225@192.168.88.102:5060>
Content-Type: application/sdp
Content-Length: 243
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="1a4c3c0c16d4f6e70264137c69ef24bd",nonce="651ddeff",algorithm=MD5

v=0
o=Cisco-SIPUA 5578 17803 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 16756 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (12 headers 11 lines) ---
Sending to 192.168.88.102 : 5060 (no NAT)
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 0.0.0.0:16756

<--- Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK1960bfd0;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
CSeq: 105 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8155054487@192.168.20.100>
Content-Length: 0


<------------>
Audio is at 192.168.20.100 port 14816
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK1960bfd0;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
CSeq: 105 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8155054487@192.168.20.100>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 525914847 525914850 IN IP4 192.168.20.100
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.20.100
t=0 0
m=audio 14816 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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

<------------>
    -- Started music on hold, class 'default', on DAHDI/1-1

<--- SIP read from UDP:192.168.88.102:50445 --->
ACK sip:8155054487@192.168.20.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK7f1df22d
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
Date: Tue, 19 Oct 2010 07:59:38 GMT
CSeq: 105 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="1a4c3c0c16d4f6e70264137c69ef24bd",nonce="651ddeff",algorithm=MD5
Content-Length: 0

<--- SIP read from UDP:192.168.88.102:50445 --->
INVITE sip:8155054487@192.168.20.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK0d583384
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
Date: Tue, 19 Oct 2010 07:59:42 GMT
CSeq: 106 INVITE
User-Agent: CSCO/7
Contact: <sip:2225@192.168.88.102:5060>
Content-Type: application/sdp
Content-Length: 249
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="1a4c3c0c16d4f6e70264137c69ef24bd",nonce="651ddeff",algorithm=MD5

v=0
o=Cisco-SIPUA 1642 3698 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 192.168.88.102
t=0 0
m=audio 16756 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (12 headers 11 lines) ---
Sending to 192.168.88.102 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK0d583384;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
CSeq: 106 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8155054487@192.168.20.100>
Content-Length: 0


<------------>
Audio is at 192.168.20.100 port 14816
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK0d583384;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
CSeq: 106 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8155054487@192.168.20.100>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 525914847 525914850 IN IP4 192.168.20.100
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.20.100
t=0 0
m=audio 14816 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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

<------------>

<--- SIP read from UDP:192.168.88.102:50445 --->
ACK sip:8155054487@192.168.20.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK2687a559
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
Date: Tue, 19 Oct 2010 07:59:42 GMT
CSeq: 106 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="1a4c3c0c16d4f6e70264137c69ef24bd",nonce="651ddeff",algorithm=MD5
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.88.102:50445 --->
INVITE sip:8155054487@192.168.20.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK3891b43a
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
Date: Tue, 19 Oct 2010 07:59:45 GMT
CSeq: 107 INVITE
User-Agent: CSCO/7
Contact: <sip:2225@192.168.88.102:5060>
Content-Type: application/sdp
Content-Length: 242
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="1a4c3c0c16d4f6e70264137c69ef24bd",nonce="651ddeff",algorithm=MD5

v=0
o=Cisco-SIPUA 7942 2759 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 16756 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (12 headers 11 lines) ---
Sending to 192.168.88.102 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK3891b43a;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
CSeq: 107 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8155054487@192.168.20.100>
Content-Length: 0


<------------>
Audio is at 192.168.20.100 port 14816
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK3891b43a;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
CSeq: 107 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8155054487@192.168.20.100>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 525914847 525914850 IN IP4 192.168.20.100
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.20.100
t=0 0
m=audio 14816 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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

<------------>

<--- SIP read from UDP:192.168.88.102:50445 --->
ACK sip:8155054487@192.168.20.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK18d0056d
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a04a6466549be-0a2c9018
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as20370cfe
Call-ID: 000ed72e-471a0006-3df33eec-70c1a379@192.168.88.102
Date: Tue, 19 Oct 2010 07:59:46 GMT
CSeq: 107 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="1a4c3c0c16d4f6e70264137c69ef24bd",nonce="651ddeff",algorithm=MD5
Content-Length: 0