All pastes #730564 Raw Edit

Miscellany

public text v1 · immutable
·published 2007-10-09 10:42 UTC
rendered paste body
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:214019@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4DFAA20E
CSeq: 2134 INVITE
To: <sip:214019@192.168.11.124>
Content-Type: application/sdp
From: <sip:214077@192.168.11.124>;tag=23DB806F
Call-ID: 547201740@192.168.11.123
Subject: sip:214077@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214077@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 32774 RTP/AVP 8 97 3 0
m=audio 32774 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 - 547201740@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=z9hG4bK4DFAA20E;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=23DB806F
To: <sip:214019@192.168.11.124>;tag=as37f1c9b2
Call-ID: 547201740@192.168.11.123
CSeq: 2134 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="606c3fbb"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '547201740@192.168.11.123' in 32000 ms (Method: INVITE)
Found user '214077'
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:214019@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4DFAA20E
CSeq: 2134 ACK
To: <sip:214019@192.168.11.124>;tag=as37f1c9b2
From: <sip:214077@192.168.11.124>;tag=23DB806F
Call-ID: 547201740@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214077@192.168.11.123;transport=udp>


<------------->
--- (9 headers 0 lines) ---
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:214019@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK2A48836A
CSeq: 2135 INVITE
To: <sip:214019@192.168.11.124>
Proxy-Authorization: Digest username="214077", realm="asterisk", nonce="606c3fbb", uri="sip:214019@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="e5a8ab65ca6f335eb27af7ed617923c1", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:214077@192.168.11.124>;tag=23DB806F
Call-ID: 547201740@192.168.11.123
Subject: sip:214077@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214077@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 32774 RTP/AVP 8 97 3 0
m=audio 32774 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 - 547201740@192.168.11.123
Found user '214077'
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 192.168.11.123:32774
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 - 0x8 (alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/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 192.168.11.123:32774
Looking for 214019 in from-sip (domain 192.168.11.124)
list_route: hop: <sip:214077@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=z9hG4bK2A48836A;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=23DB806F
To: <sip:214019@192.168.11.124>
Call-ID: 547201740@192.168.11.123
CSeq: 2135 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214019@192.168.11.124>
Content-Length: 0


<------------>
Audio is at 192.168.11.124 port 15878
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.11.120:5060:
INVITE sip:214019@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240f8219;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as56c918c2
To: <sip:214019@192.168.11.120>
Contact: <sip:214077@192.168.11.124>
Call-ID: 4626a4955f6eb78915d45c1922f57057@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 09 Oct 2007 10:41:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 522

v=0
o=root 21640 21640 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 15878 RTP/AVP 8 3 0 112 5 10 7 18 97 111 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
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=z9hG4bK240f8219;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as56c918c2
To: <sip:214019@192.168.11.120>;tag=DE34962C-14F4
Date: Tue, 09 Oct 2007 10:40:32 GMT
Call-ID: 4626a4955f6eb78915d45c1922f57057@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/2.0 486 Busy here
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240f8219;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as56c918c2
To: <sip:214019@192.168.11.120>;tag=DE34962C-14F4
Date: Tue, 09 Oct 2007 10:40:32 GMT
Call-ID: 4626a4955f6eb78915d45c1922f57057@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:214019@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240f8219;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as56c918c2
To: <sip:214019@192.168.11.120>;tag=DE34962C-14F4
Contact: <sip:214077@192.168.11.124>
Call-ID: 4626a4955f6eb78915d45c1922f57057@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
bee*CLI> 
<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK2A48836A;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=23DB806F
To: <sip:214019@192.168.11.124>;tag=as1af642f3
Call-ID: 547201740@192.168.11.123
CSeq: 2135 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214019@192.168.11.124>
Content-Length: 0
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17


<------------>
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:214019@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK2A48836A
CSeq: 2135 ACK
To: <sip:214019@192.168.11.124>;tag=as1af642f3
From: <sip:214077@192.168.11.124>;tag=23DB806F
Call-ID: 547201740@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214077@192.168.11.123;transport=udp>


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '4626a4955f6eb78915d45c1922f57057@192.168.11.124' Method: INVITE
Really destroying SIP dialog '547201740@192.168.11.123' Method: ACK
Really destroying SIP dialog '1877302052@192.168.11.123' Method: REGISTER
bee*CLI> 
<--- SIP read from 192.168.11.120:56021 --->
BYE sip:216@192.168.11.124:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.11.120:5060
From: <sip:9511804@192.168.11.120>;tag=DE32DA63-10E6
To: "216" <sip:216@192.168.11.124>;tag=as1d1d919c
Date: Tue, 09 Oct 2007 10:38:45 GMT
Call-ID: 29d964f528d6bfb678c6964a2e37bd47@192.168.11.124
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 6
Timestamp: 1191926437
CSeq: 101 BYE
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.11.120 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.11.120:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP  192.168.11.120:5060;received=192.168.11.120
From: <sip:9511804@192.168.11.120>;tag=DE32DA63-10E6
To: "216" <sip:216@192.168.11.124>;tag=as1d1d919c
Call-ID: 29d964f528d6bfb678c6964a2e37bd47@192.168.11.124
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:216@192.168.11.124>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'D1B9-1C94-47111489E1C3E56C1892-222@SipHost' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:216@192.168.11.131:5060> for address/port to send to
set_destination: set destination to 192.168.11.131, port 5060
Reliably Transmitting (no NAT) to 192.168.11.131:5060:
BYE sip:216@192.168.11.131:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK0809cd3a;rport
From: <sip:9511804@195.239.214.91:5060;user=phone>;tag=as2381c048
To: "216" <sip:216@195.239.214.91:5060>;tag=8f34d32b-111489
Call-ID: D1B9-1C94-47111489E1C3E56C1892-222@SipHost
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
bee*CLI> 
<--- SIP read from 192.168.11.131:5060 --->
SIP/2.0 200 OK
Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK0809cd3a
From:<sip:9511804@195.239.214.91:5060;user=phone>;tag=as2381c048
To:"216" <sip:216@195.239.214.91:5060>;tag=8f34d32b-111489
Call-ID:D1B9-1C94-47111489E1C3E56C1892-222@SipHost
CSeq:102 BYE
User-Agent:dlink 12-36-7924821
Content-Length:0


<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '29d964f528d6bfb678c6964a2e37bd47@192.168.11.124' Method: BYE
Really destroying SIP dialog 'D1B9-1C94-47111489E1C3E56C1892-222@SipHost' Method: ACK
bee*CLI> 
<--- SIP read from 192.168.11.131:5060 --->
BYE sip:9511804@192.168.11.124 SIP/2.0
Via:SIP/2.0/UDP 192.168.11.131:5060;branch=z9hG4bK23964c042b2642e6
From:"216" <sip:216@195.239.214.91:5060>;tag=8f34d32b-111489
To:<sip:9511804@195.239.214.91:5060;user=phone>;tag=as2381c048
Call-ID:D1B9-1C94-47111489E1C3E56C1892-222@SipHost
CSeq:6018 BYE
Max-Forwards:70
User-Agent:dlink 12-36-7924821
Content-Length:0


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

<--- Transmitting (no NAT) to 192.168.11.131:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.11.131:5060;branch=z9hG4bK23964c042b2642e6;received=192.168.11.131
From: "216" <sip:216@195.239.214.91:5060>;tag=8f34d32b-111489
To: <sip:9511804@195.239.214.91:5060;user=phone>;tag=as2381c048
Call-ID: D1B9-1C94-47111489E1C3E56C1892-222@SipHost
CSeq: 6018 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Really destroying SIP dialog 'D1B9-1C94-466848152998B392D344-001@SipHost' Method: REGISTER
Really destroying SIP dialog 'D1B9-1C94-466848153EC9999E4BE8-002@SipHost' Method: REGISTER
bee*CLI> sip no debug
SIP Debugging Disabled
bee*CLI> sip set debug
SIP Debugging enabled
Really destroying SIP dialog 'D1B9-1C94-46684815AD3CE939E424-003@SipHost' Method: REGISTER
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:214019@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK578BC651
CSeq: 7379 INVITE
To: <sip:214019@192.168.11.124>
Content-Type: application/sdp
From: <sip:214077@192.168.11.124>;tag=44166B53
Call-ID: 1961359474@192.168.11.123
Subject: sip:214077@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214077@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 32776 RTP/AVP 8 97 3 0
m=audio 32776 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 - 1961359474@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=z9hG4bK578BC651;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=44166B53
To: <sip:214019@192.168.11.124>;tag=as6b735b82
Call-ID: 1961359474@192.168.11.123
CSeq: 7379 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="321e36d2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1961359474@192.168.11.123' in 32000 ms (Method: INVITE)
Found user '214077'
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:214019@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK578BC651
CSeq: 7379 ACK
To: <sip:214019@192.168.11.124>;tag=as6b735b82
From: <sip:214077@192.168.11.124>;tag=44166B53
Call-ID: 1961359474@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214077@192.168.11.123;transport=udp>


<------------->
--- (9 headers 0 lines) ---
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:214019@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4
CSeq: 7380 INVITE
To: <sip:214019@192.168.11.124>
Proxy-Authorization: Digest username="214077", realm="asterisk", nonce="321e36d2", uri="sip:214019@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="72170c1418da887224977f52900e0c28", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:214077@192.168.11.124>;tag=44166B53
Call-ID: 1961359474@192.168.11.123
Subject: sip:214077@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214077@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 32776 RTP/AVP 8 97 3 0
m=audio 32776 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 - 1961359474@192.168.11.123
Found user '214077'
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 192.168.11.123:32776
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 - 0x8 (alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/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 192.168.11.123:32776
Looking for 214019 in from-sip (domain 192.168.11.124)
list_route: hop: <sip:214077@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=z9hG4bK46FF98B4;received=192.168.11.123
rom: <sip:214077@192.168.11.124>;tag=44166B53
To: <sip:214019@192.168.11.124>
Call-ID: 1961359474@192.168.11.123
CSeq: 7380 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214019@192.168.11.124>
Content-Length: 0


<------------>
Audio is at 192.168.11.124 port 12030
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.11.120:5060:
INVITE sip:214019@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240a0997;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as5289d0d2
To: <sip:214019@192.168.11.120>
Contact: <sip:214077@192.168.11.124>
Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 09 Oct 2007 10:41:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 522

v=0
o=root 21640 21640 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 12030 RTP/AVP 8 3 0 112 5 10 7 18 97 111 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
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=z9hG4bK240a0997;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as5289d0d2
To: <sip:214019@192.168.11.120>;tag=DE34DCED-2550
Date: Tue, 09 Oct 2007 10:40:50 GMT
Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@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=z9hG4bK240a0997;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as5289d0d2
To: <sip:214019@192.168.11.120>;tag=DE34DCED-2550
Date: Tue, 09 Oct 2007 10:40:50 GMT
Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:214019@192.168.11.120:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 252

v=0
o=CiscoSystemsSIP-GW-UserAgent 8301 156 IN IP4 192.168.11.120
s=SIP Call
c=IN IP4 192.168.11.120
t=0 0
m=audio 18898 RTP/AVP 8 101
c=IN IP4 192.168.11.120
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (13 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.11.120:18898
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.11.120:18898
Audio is at 192.168.11.124 port 16230
Adding codec 0x8 (alaw) 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=z9hG4bK46FF98B4;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=44166B53
To: <sip:214019@192.168.11.124>;tag=as0353e318
Call-ID: 1961359474@192.168.11.123
CSeq: 7380 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214019@192.168.11.124>
Content-Type: application/sdp
Content-Length: 188

v=0
o=root 21640 21640 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 16230 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Really destroying SIP dialog 'D1B9-1C94-4668481595DDF37A5E3A-004@SipHost' Method: REGISTER
bee*CLI> 
<--- SIP read from 192.168.11.128:58559 --->
INVITE sip:1001@192.168.11.124:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.11.128:5060
From: "unknown" <sip:192.168.11.128>;tag=197FC956-133B
To: <sip:1001@192.168.11.124>
Date: Fri, 05 Mar 1993 22:50:04 GMT
Call-ID: 976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 2539994139-416420300-2150403679-25483487
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: "unknown" <sip:192.168.11.128>;party=calling;screen=no;privacy=off
Timestamp: 731371804
Contact: <sip:192.168.11.128:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 287

v=0
o=CiscoSystemsSIP-GW-UserAgent 3633 652 IN IP4 192.168.11.128
s=SIP Call
c=IN IP4 192.168.11.128
t=0 0
m=audio 19016 RTP/AVP 18 8 101
c=IN IP4 192.168.11.128
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

<------------->
--- (20 headers 12 lines) ---
Sending to 192.168.11.128 : 5060 (no NAT)
Using INVITE request as basis request - 976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128
Found no matching peer or user for '192.168.11.128:58559'
[Oct  9 14:41:41] NOTICE[21665]: chan_sip.c:13587 handle_request_invite: Failed to authenticate user "unknown" <sip:192.168.11.128>;tag=197FC956-133B

<--- Reliably Transmitting (no NAT) to 192.168.11.128:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP  192.168.11.128:5060;received=192.168.11.128
From: "unknown" <sip:192.168.11.128>;tag=197FC956-133B
To: <sip:1001@192.168.11.124>;tag=as1aed72aa
Call-ID: 976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128' in 32000 ms (Method: INVITE)
bee*CLI> 
<--- SIP read from 192.168.11.128:58559 --->
ACK sip:1001@192.168.11.124:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.11.128:5060
From: "unknown" <sip:192.168.11.128>;tag=197FC956-133B
To: <sip:1001@192.168.11.124>;tag=as1aed72aa
Date: Fri, 05 Mar 1993 22:50:04 GMT
Call-ID: 976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128' Method: ACK
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
CANCEL sip:214019@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4
CSeq: 7380 CANCEL
To: <sip:214019@192.168.11.124>
From: <sip:214077@192.168.11.124>;tag=44166B53
Call-ID: 1961359474@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214077@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=z9hG4bK46FF98B4;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=44166B53
To: <sip:214019@192.168.11.124>;tag=as0353e318
Call-ID: 1961359474@192.168.11.123
CSeq: 7380 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=z9hG4bK46FF98B4;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=44166B53
To: <sip:214019@192.168.11.124>;tag=as0353e318
Call-ID: 1961359474@192.168.11.123
CSeq: 7380 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214019@192.168.11.124>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.11.120:5060:
CANCEL sip:214019@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240a0997;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as5289d0d2
To: <sip:214019@192.168.11.120>
Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '2efa2dfe12e18f220b3f11c47102be7f@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=z9hG4bK240a0997;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as5289d0d2
To: <sip:214019@192.168.11.120>
Date: Tue, 09 Oct 2007 10:40:54 GMT
Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124
Content-Length: 0
CSeq: 102 CANCEL


<------------->
--- (8 headers 0 lines) ---
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=z9hG4bK240a0997;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as5289d0d2
To: <sip:214019@192.168.11.120>;tag=DE34DCED-2550
Date: Tue, 09 Oct 2007 10:40:54 GMT
Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@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:214019@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240a0997;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as5289d0d2
To: <sip:214019@192.168.11.120>;tag=DE34DCED-2550
Contact: <sip:214077@192.168.11.124>
Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Really destroying SIP dialog '2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124' Method: INVITE
bee*CLI> 
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:214019@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4
CSeq: 7380 ACK
From: <sip:214077@192.168.11.124>;tag=44166B53
To: <sip:214019@192.168.11.124>;tag=as0353e318
Call-ID: 1961359474@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214077@192.168.11.123;transport=udp>


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1961359474@192.168.11.123' Method: ACK
bee*CLI> sip no debug
SIP Debugging Disabled
bee*CLI>