<--- SIP read from UDP:192.168.88.102:50445 --->
INVITE sip:8155054487@192.168.20.100;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK6e5133cc
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:12 GMT
CSeq: 101 INVITE
User-Agent: CSCO/7
Contact: <sip:2225@192.168.88.102:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 250
Accept: application/sdp

v=0
o=Cisco-SIPUA 8604 25816 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 192.168.88.102
t=0 0
m=audio 16758 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

<------------->
--- (13 headers 11 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 192.168.88.102 : 5060 (no NAT)
Using INVITE request as basis request - 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Found peer '2225' for '2225' from 192.168.88.102:50445

<--- Reliably Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK6e5133cc;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as0282b068
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2614f3f1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.88.102:52542 --->
ACK sip:8155054487@192.168.20.100;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK6e5133cc
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as0282b068
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:12 GMT
CSeq: 101 ACK
Content-Length: 0


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

<--- SIP read from UDP:192.168.88.102:50445 --->
INVITE sip:8155054487@192.168.20.100;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK11c4ba5d
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:13 GMT
CSeq: 102 INVITE
User-Agent: CSCO/7
Contact: <sip:2225@192.168.88.102:5060>
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5
Expires: 180
Content-Type: application/sdp
Content-Length: 250

v=0
o=Cisco-SIPUA 8604 25816 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 192.168.88.102
t=0 0
m=audio 16758 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

<------------->
--- (13 headers 11 lines) ---
Sending to 192.168.88.102 : 5060 (no NAT)
Using INVITE request as basis request - 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Found peer '2225' for '2225' from 192.168.88.102:50445
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 192.168.88.102:16758
Looking for 8155054487 in rri (domain 192.168.20.100)
list_route: hop: <sip:2225@192.168.88.102:5060>

<--- 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=z9hG4bK11c4ba5d;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 102 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


<------------>
    -- Executing [8155054487@rri:1] Goto("SIP/2225-00000001", "18155054487,1") in new stack
    -- Goto (rri,18155054487,1)
    -- Executing [18155054487@rri:1] Macro("SIP/2225-00000001", "localcall,18155054487") in new stack
    -- Executing [s@macro-localcall:1] Set("SIP/2225-00000001", "dialMacro1=dialmcleod") in new stack
    -- Executing [s@macro-localcall:2] Set("SIP/2225-00000001", "dialMacro2=dialtnt") in new stack
    -- Executing [s@macro-localcall:3] Macro("SIP/2225-00000001", "dialfromvariables,18155054487,120") in new stack
    -- Executing [s@macro-dialfromvariables:1] Macro("SIP/2225-00000001", "setCID,2225,rri") in new stack
    -- Executing [s@macro-setCID:1] NoOp("SIP/2225-00000001", "Caller ID has 4 digits") in new stack
    -- Executing [s@macro-setCID:2] GotoIf("SIP/2225-00000001", "0?100:3") in new stack
    -- Goto (macro-setCID,s,3)
    -- Executing [s@macro-setCID:3] Set("SIP/2225-00000001", "newCID=8155054487") in new stack
    -- Executing [s@macro-setCID:4] Set("SIP/2225-00000001", "newCIDNAME=") in new stack
    -- Executing [s@macro-setCID:5] GotoIf("SIP/2225-00000001", "1?6:20") in new stack
    -- Goto (macro-setCID,s,6)
    -- Executing [s@macro-setCID:6] GotoIf("SIP/2225-00000001", "0?40:30") in new stack
    -- Goto (macro-setCID,s,30)
    -- Executing [s@macro-setCID:30] Set("SIP/2225-00000001", "newCIDNAME=") in new stack
    -- Executing [s@macro-setCID:31] Goto("SIP/2225-00000001", "40") in new stack
    -- Goto (macro-setCID,s,40)
    -- Executing [s@macro-setCID:40] NoOp("SIP/2225-00000001", "Setting New Caller ID Number: 8155054487") in new stack
    -- Executing [s@macro-setCID:41] Set("SIP/2225-00000001", "CALLERID(number)=8155054487") in new stack
    -- Executing [s@macro-setCID:42] Set("SIP/2225-00000001", "CALLERID(ANI)=8155054487") in new stack
    -- Executing [s@macro-setCID:43] NoOp("SIP/2225-00000001", "Setting New Caller ID Name: ") in new stack
    -- Executing [s@macro-setCID:44] Set("SIP/2225-00000001", "CALLERID(name)=") in new stack
    -- Executing [s@macro-dialfromvariables:2] Set("SIP/2225-00000001", "dialOrder=0") in new stack
    -- Executing [s@macro-dialfromvariables:3] Goto("SIP/2225-00000001", "100") in new stack
    -- Goto (macro-dialfromvariables,s,100)
    -- Executing [s@macro-dialfromvariables:100] Set("SIP/2225-00000001", "dialOrder=1") in new stack
    -- Executing [s@macro-dialfromvariables:101] GotoIf("SIP/2225-00000001", "1?102:200") in new stack
    -- Goto (macro-dialfromvariables,s,102)
    -- Executing [s@macro-dialfromvariables:102] NoOp("SIP/2225-00000001", "dialmcleod is the macro") in new stack
    -- Executing [s@macro-dialfromvariables:103] Macro("SIP/2225-00000001", "dialmcleod,18155054487,120") in new stack
    -- Executing [s@macro-dialmcleod:1] Set("SIP/2225-00000001", "CDR(userfield)=Outgoing PRI via McCleod") in new stack
    -- Executing [s@macro-dialmcleod:2] Dial("SIP/2225-00000001", "DAHDI/g1/18155054487,120") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g1/18155054487
    -- DAHDI/1-1 is proceeding passing it to SIP/2225-00000001

<--- 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=z9hG4bK11c4ba5d;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 102 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

    -- DAHDI/1-1 is making progress passing it to SIP/2225-00000001
Audio is at 192.168.20.100 port 15518
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK11c4ba5d;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 102 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 774624382 774624382 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 15518 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

<------------>
    -- DAHDI/1-1 is making progress passing it to SIP/2225-00000001
    -- DAHDI/1-1 is making progress passing it to SIP/2225-00000001
    -- DAHDI/1-1 is making progress passing it to SIP/2225-00000001
    -- DAHDI/1-1 is ringing

<--- Transmitting (no NAT) to 192.168.88.102:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.88.102:5060;branch=z9hG4bK11c4ba5d;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 102 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

    -- DAHDI/1-1 answered SIP/2225-00000001
Audio is at 192.168.20.100 port 15518
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=z9hG4bK11c4ba5d;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 102 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 774624382 774624383 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 15518 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=z9hG4bK0eed21d8
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:30 GMT
CSeq: 102 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",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=z9hG4bK0e53b34f
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:36 GMT
CSeq: 103 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="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5

v=0
o=Cisco-SIPUA 7325 3116 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 16758 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=z9hG4bK0e53b34f;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@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 15518
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=z9hG4bK0e53b34f;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@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 774624382 774624383 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 15518 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=z9hG4bK590ae2a5
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:36 GMT
CSeq: 103 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",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=z9hG4bK67f429d2
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:38 GMT
CSeq: 104 INVITE
User-Agent: CSCO/7
Contact: <sip:2225@192.168.88.102:5060>
Content-Type: application/sdp
Content-Length: 250
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5

v=0
o=Cisco-SIPUA 12717 1738 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 192.168.88.102
t=0 0
m=audio 16758 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=z9hG4bK67f429d2;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@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 15518
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=z9hG4bK67f429d2;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@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 774624382 774624383 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 15518 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=z9hG4bK5c943a69
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:39 GMT
CSeq: 104 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",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=z9hG4bK0d422be3
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:41 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="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5

v=0
o=Cisco-SIPUA 1195 28397 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 16758 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 0.0.0.0:16758

<--- 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=z9hG4bK0d422be3;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@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 15518
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=z9hG4bK0d422be3;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@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 774624382 774624384 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 15518 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=z9hG4bK504ccf43
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:41 GMT
CSeq: 105 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",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=z9hG4bK54489105
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:45 GMT
CSeq: 106 INVITE
User-Agent: CSCO/7
Contact: <sip:2225@192.168.88.102:5060>
Content-Type: application/sdp
Content-Length: 250
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5

v=0
o=Cisco-SIPUA 12814 1776 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 192.168.88.102
t=0 0
m=audio 16758 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=z9hG4bK54489105;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@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 15518
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=z9hG4bK54489105;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@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 774624382 774624384 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 15518 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=z9hG4bK7c36ba81
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:45 GMT
CSeq: 106 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",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=z9hG4bK184c976b
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:48 GMT
CSeq: 107 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="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5

v=0
o=Cisco-SIPUA 13340 27205 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 16758 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=z9hG4bK184c976b;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@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 15518
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=z9hG4bK184c976b;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@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 774624382 774624384 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 15518 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=z9hG4bK526d7131
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:49 GMT
CSeq: 107 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",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=z9hG4bK201ab9f8
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:51 GMT
CSeq: 108 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="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5

v=0
o=Cisco-SIPUA 14745 14201 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 192.168.88.102
t=0 0
m=audio 16758 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=z9hG4bK201ab9f8;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 108 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 15518
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=z9hG4bK201ab9f8;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 108 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 774624382 774624384 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 15518 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=z9hG4bK6c5e88aa
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:51 GMT
CSeq: 108 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",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=z9hG4bK181bb56d
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:55 GMT
CSeq: 109 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="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5

v=0
o=Cisco-SIPUA 21119 12287 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 16758 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=z9hG4bK181bb56d;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 109 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 15518
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=z9hG4bK181bb56d;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 109 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 774624382 774624384 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 15518 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=z9hG4bK22231b1b
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:40:55 GMT
CSeq: 109 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",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=z9hG4bK2ba12aa8
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:41:00 GMT
CSeq: 110 INVITE
User-Agent: CSCO/7
Contact: <sip:2225@192.168.88.102:5060>
Content-Type: application/sdp
Content-Length: 250
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5

v=0
o=Cisco-SIPUA 17266 1403 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 192.168.88.102
t=0 0
m=audio 16758 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=z9hG4bK2ba12aa8;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 110 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 15518
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=z9hG4bK2ba12aa8;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 110 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 774624382 774624384 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 15518 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=z9hG4bK2bbba460
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:41:00 GMT
CSeq: 110 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",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=z9hG4bK701a01a3
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:41:04 GMT
CSeq: 111 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="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5

v=0
o=Cisco-SIPUA 19035 14009 IN IP4 192.168.88.102
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 16758 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=z9hG4bK701a01a3;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 111 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 15518
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=z9hG4bK701a01a3;received=192.168.88.102
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
CSeq: 111 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 774624382 774624384 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 15518 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=z9hG4bK38d42244
From: "2225" <sip:2225@192.168.20.100>;tag=000ed72e471a082800349331-399ac16e
To: <sip:8155054487@192.168.20.100;user=phone>;tag=as171f3210
Call-ID: 000ed72e-471a0007-14108a06-263d9dbd@192.168.88.102
Date: Tue, 19 Oct 2010 08:41:04 GMT
CSeq: 111 ACK
User-Agent: CSCO/7
Authorization: Digest username="2225",realm="asterisk",uri="sip:192.168.20.100",response="8ff8f18c05519743f294d2ce9632e42c",nonce="2614f3f1",algorithm=MD5
Content-Length: 0