Public feed #721631

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·published 2007-10-01 12:55 UTC
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<------------>
Scheduling destruction of SIP dialog '7ae049f51b17b2d930ded27474cad6bc@192.168.11.129' in 32000 ms (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=z9hG4bK4469F3DD
CSeq: 3872 INVITE
To: <sip:214019@192.168.11.124>
Content-Type: application/sdp
From: <sip:214077@192.168.11.124>;tag=225F6D3F
Call-ID: 250702825@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 32804 RTP/AVP 8 97 3 0
m=audio 32804 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 - 250702825@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=z9hG4bK4469F3DD;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=225F6D3F
To: <sip:214019@192.168.11.124>;tag=as233d372d
Call-ID: 250702825@192.168.11.123
CSeq: 3872 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a2952bc"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '250702825@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=z9hG4bK4469F3DD
CSeq: 3872 ACK
To: <sip:214019@192.168.11.124>;tag=as233d372d
From: <sip:214077@192.168.11.124>;tag=225F6D3F
Call-ID: 250702825@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=z9hG4bK5CE8F01F
CSeq: 3873 INVITE
To: <sip:214019@192.168.11.124>
Proxy-Authorization: Digest username="214077", realm="asterisk", nonce="3a2952bc", uri="sip:214019@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="f3424eddcd6b8e09480b32fd8225b8d6", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:214077@192.168.11.124>;tag=225F6D3F
Call-ID: 250702825@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 32804 RTP/AVP 8 97 3 0
m=audio 32804 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 - 250702825@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 192.168.11.123:32804
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 192.168.11.123:32804
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=z9hG4bK5CE8F01F;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=225F6D3F
To: <sip:214019@192.168.11.124>
Call-ID: 250702825@192.168.11.123
CSeq: 3873 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 13684
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=z9hG4bK5f1284ea;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as03a357bf
To: <sip:214019@192.168.11.120>
Contact: <sip:214077@192.168.11.124>
Call-ID: 392f8c2d588756633723704446b7fbd2@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Oct 2007 12:54:33 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 13684 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=z9hG4bK5f1284ea;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as03a357bf
To: <sip:214019@192.168.11.120>;tag=B57BCEB1-440
Date: Mon, 01 Oct 2007 12:53:48 GMT
Call-ID: 392f8c2d588756633723704446b7fbd2@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=z9hG4bK5f1284ea;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as03a357bf
To: <sip:214019@192.168.11.120>;tag=B57BCEB1-440
Date: Mon, 01 Oct 2007 12:53:48 GMT
Call-ID: 392f8c2d588756633723704446b7fbd2@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: 253

v=0
o=CiscoSystemsSIP-GW-UserAgent 1454 1359 IN IP4 192.168.11.120
s=SIP Call
c=IN IP4 192.168.11.120
t=0 0
m=audio 18412 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 192.168.11.120:18412
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 192.168.11.120:18412
Audio is at 192.168.11.124 port 16638
Adding codec 0x8 (alaw) to SDP
bee*CLI> 
<--- 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=z9hG4bK5CE8F01F;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=225F6D3F
To: <sip:214019@192.168.11.124>;tag=as4b563b6c
Call-ID: 250702825@192.168.11.123
CSeq: 3873 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 16638 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
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=z9hG4bK5CE8F01F
CSeq: 3873 CANCEL
To: <sip:214019@192.168.11.124>
From: <sip:214077@192.168.11.124>;tag=225F6D3F
Call-ID: 250702825@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=z9hG4bK5CE8F01F;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=225F6D3F
To: <sip:214019@192.168.11.124>;tag=as4b563b6c
Call-ID: 250702825@192.168.11.123
CSeq: 3873 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=z9hG4bK5CE8F01F;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=225F6D3F
To: <sip:214019@192.168.11.124>;tag=as4b563b6c
Call-ID: 250702825@192.168.11.123
CSeq: 3873 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 '392f8c2d588756633723704446b7fbd2@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=z9hG4bK5f1284ea;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as03a357bf
To: <sip:214019@192.168.11.120>
Call-ID: 392f8c2d588756633723704446b7fbd2@192.168.11.124
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '392f8c2d588756633723704446b7fbd2@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=z9hG4bK5f1284ea;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as03a357bf
To: <sip:214019@192.168.11.120>
Date: Mon, 01 Oct 2007 12:53:50 GMT
Call-ID: 392f8c2d588756633723704446b7fbd2@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=z9hG4bK5f1284ea;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as03a357bf
To: <sip:214019@192.168.11.120>;tag=B57BCEB1-440
Date: Mon, 01 Oct 2007 12:53:50 GMT
Call-ID: 392f8c2d588756633723704446b7fbd2@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=z9hG4bK5f1284ea;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as03a357bf
To: <sip:214019@192.168.11.120>;tag=B57BCEB1-440
Contact: <sip:214077@192.168.11.124>
Call-ID: 392f8c2d588756633723704446b7fbd2@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Really destroying SIP dialog '392f8c2d588756633723704446b7fbd2@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=z9hG4bK5CE8F01F
CSeq: 3873 ACK
From: <sip:214077@192.168.11.124>;tag=225F6D3F
To: <sip:214019@192.168.11.124>;tag=as4b563b6c
Call-ID: 250702825@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214077@192.168.11.123;transport=udp>


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