<--- SIP read from 192.168.11.123:5060 --->
REGISTER sip:192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK3C152F56
CSeq: 2471 REGISTER
To: <sip:214058@192.168.11.124>
Expires: 900
From: <sip:214058@192.168.11.124>
Call-ID: 2059936848@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Event: registration
Allow-Events: presence
Contact: <sip:214058@192.168.11.123;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER"


<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.11.123 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK3C152F56;received=192.168.11.123
From: <sip:214058@192.168.11.124>
To: <sip:214058@192.168.11.124>
Call-ID: 2059936848@192.168.11.123
CSeq: 2471 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214058@192.168.11.124>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK3C152F56;received=192.168.11.123
From: <sip:214058@192.168.11.124>
To: <sip:214058@192.168.11.124>;tag=as6315da0f
Call-ID: 2059936848@192.168.11.123
CSeq: 2471 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a1d1be3"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2059936848@192.168.11.123' in 32000 ms (Method: REGISTER)
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
REGISTER sip:192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK10523F7E
CSeq: 2472 REGISTER
To: <sip:214058@192.168.11.124>
Authorization: Digest username="214058", realm="asterisk", nonce="3a1d1be3", uri="sip:192.168.11.124", cnonce="abcdefghi", nc=00000001, response="4924f83eacd60eb4af4d9f8585a71b34", opaque="", algorithm="MD5"
Expires: 900
From: <sip:214058@192.168.11.124>
Call-ID: 2059936848@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Event: registration
Allow-Events: presence
Contact: <sip:214058@192.168.11.123;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER"


<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.11.123 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK10523F7E;received=192.168.11.123
From: <sip:214058@192.168.11.124>
To: <sip:214058@192.168.11.124>
Call-ID: 2059936848@192.168.11.123
CSeq: 2472 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214058@192.168.11.124>
Content-Length: 0


<------------>
[Jul 10 22:17:16] WARNING[8183]: db.c:66 dbinit: Unable to open Asterisk database

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK10523F7E;received=192.168.11.123
From: <sip:214058@192.168.11.124>
To: <sip:214058@192.168.11.124>;tag=as6315da0f
Call-ID: 2059936848@192.168.11.123
CSeq: 2472 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 900
Contact: <sip:214058@192.168.11.123;transport=udp>;expires=900
Date: Tue, 10 Jul 2007 18:17:16 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2059936848@192.168.11.123' in 32000 ms (Method: REGISTER)
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK12C442BB
CSeq: 703 INVITE
To: <sip:9214020@192.168.11.124>
Content-Type: application/sdp
From: <sip:214058@192.168.11.124>;tag=7DDD2ABF
Call-ID: 580173408@192.168.11.123
Subject: sip:214058@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.11.123
s=The Funky Flow
c=IN IP4 192.168.11.123
t=0 0
m=audio 32924 RTP/AVP 8 97 3 0
m=audio 32924 RTP/AVP 97 3 0
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30

<------------->
--- (11 headers 12 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 580173408@192.168.11.123

<--- Reliably Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK12C442BB;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=7DDD2ABF
To: <sip:9214020@192.168.11.124>;tag=as5c331205
Call-ID: 580173408@192.168.11.123
CSeq: 703 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a4910b7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '580173408@192.168.11.123' in 32000 ms (Method: INVITE)
Found user '214058'
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK12C442BB
CSeq: 703 ACK
To: <sip:9214020@192.168.11.124>;tag=as5c331205
From: <sip:214058@192.168.11.124>;tag=7DDD2ABF
Call-ID: 580173408@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>


<------------->
--- (9 headers 0 lines) ---
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK65B842E1
CSeq: 704 INVITE
To: <sip:9214020@192.168.11.124>
Proxy-Authorization: Digest username="214058", realm="asterisk", nonce="4a4910b7", uri="sip:9214020@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="cd1451b6c222e1b591f77d4774cdb255", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:214058@192.168.11.124>;tag=7DDD2ABF
Call-ID: 580173408@192.168.11.123
Subject: sip:214058@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.11.123
s=The Funky Flow
c=IN IP4 192.168.11.123
t=0 0
m=audio 32924 RTP/AVP 8 97 3 0
m=audio 32924 RTP/AVP 97 3 0
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30

<------------->
--- (12 headers 12 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 580173408@192.168.11.123
Found user '214058'
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Peer audio RTP is at port 192.168.11.123:32924
Found description format PCMU for ID 0
Found description format GSM for ID 3
Found description format PCMA for ID 8
Found description format iLBC for ID 97
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.11.123:32924
Looking for 9214020 in local-phones (domain 192.168.11.124)
list_route: hop: <sip:214058@192.168.11.123;transport=udp>

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK65B842E1;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=7DDD2ABF
To: <sip:9214020@192.168.11.124>
Call-ID: 580173408@192.168.11.123
CSeq: 704 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9214020@192.168.11.124>
Content-Length: 0


<------------>
Audio is at 192.168.11.124 port 17964
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.11.120:5060:
INVITE sip:214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK15d8d8d9;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as412b295d
To: <sip:214020@192.168.11.120>
Contact: <sip:214058@192.168.11.124>
Call-ID: 51d2777a0982a2734fac840d4bd5120a@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 10 Jul 2007 18:17:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 8154 8154 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 17964 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
bee*CLI> 
<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK15d8d8d9;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as412b295d
To: <sip:214020@192.168.11.120>;tag=A453869-AEE
Date: Tue, 10 Jul 2007 18:17:20 GMT
Call-ID: 51d2777a0982a2734fac840d4bd5120a@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
bee*CLI> 
<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK15d8d8d9;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as412b295d
To: <sip:214020@192.168.11.120>;tag=A453869-AEE
Date: Tue, 10 Jul 2007 18:17:20 GMT
Call-ID: 51d2777a0982a2734fac840d4bd5120a@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:214020@192.168.11.120:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 197

v=0
o=CiscoSystemsSIP-GW-UserAgent 6554 6392 IN IP4 192.168.11.120
s=SIP Call
c=IN IP4 192.168.11.120
t=0 0
m=audio 19202 RTP/AVP 8
c=IN IP4 192.168.11.120
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (13 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 192.168.11.120:19202
Found description format PCMA for ID 8
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.11.120:19202
Audio is at 192.168.11.124 port 16138
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK65B842E1;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=7DDD2ABF
To: <sip:9214020@192.168.11.124>;tag=as5b35b203
Call-ID: 580173408@192.168.11.123
CSeq: 704 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9214020@192.168.11.124>
Content-Type: application/sdp
Content-Length: 233

v=0
o=root 8154 8154 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 16138 RTP/AVP 8 3 0
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
CANCEL sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK65B842E1
CSeq: 704 CANCEL
To: <sip:9214020@192.168.11.124>
From: <sip:214058@192.168.11.124>;tag=7DDD2ABF
Call-ID: 580173408@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>


<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK65B842E1;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=7DDD2ABF
To: <sip:9214020@192.168.11.124>;tag=as5b35b203
Call-ID: 580173408@192.168.11.123
CSeq: 704 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK65B842E1;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=7DDD2ABF
To: <sip:9214020@192.168.11.124>;tag=as5b35b203
Call-ID: 580173408@192.168.11.123
CSeq: 704 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9214020@192.168.11.124>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '51d2777a0982a2734fac840d4bd5120a@192.168.11.124' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.11.120:5060:
CANCEL sip:214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK15d8d8d9;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as412b295d
To: <sip:214020@192.168.11.120>
Call-ID: 51d2777a0982a2734fac840d4bd5120a@192.168.11.124
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '51d2777a0982a2734fac840d4bd5120a@192.168.11.124' in 32000 ms (Method: INVITE)
bee*CLI> 
<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK15d8d8d9;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as412b295d
To: <sip:214020@192.168.11.120>
Date: Tue, 10 Jul 2007 18:17:24 GMT
Call-ID: 51d2777a0982a2734fac840d4bd5120a@192.168.11.124
Content-Length: 0
CSeq: 102 CANCEL


<------------->
--- (8 headers 0 lines) ---
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK65B842E1
CSeq: 704 ACK
From: <sip:214058@192.168.11.124>;tag=7DDD2ABF
To: <sip:9214020@192.168.11.124>;tag=as5b35b203
Call-ID: 580173408@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '580173408@192.168.11.123' Method: ACK
bee*CLI> 
<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK15d8d8d9;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as412b295d
To: <sip:214020@192.168.11.120>;tag=A453869-AEE
Date: Tue, 10 Jul 2007 18:17:24 GMT
Call-ID: 51d2777a0982a2734fac840d4bd5120a@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.11.120:5060:
ACK sip:214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK15d8d8d9;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as412b295d
To: <sip:214020@192.168.11.120>;tag=A453869-AEE
Contact: <sip:214058@192.168.11.124>
Call-ID: 51d2777a0982a2734fac840d4bd5120a@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---