rendered paste body#
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
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From: "214077" <sip:214077@192.168.11.124>;tag=as5289d0d2
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To: <sip:214019@192.168.11.120>;tag=DE34DCED-2550
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Contact: <sip:214077@192.168.11.124>
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Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124
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CSeq: 102 ACK
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User-Agent: Asterisk PBX
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Max-Forwards: 70
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Content-Length: 0
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---
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Really destroying SIP dialog '2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124' Method: INVITE
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bee*CLI>
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<--- SIP read from 192.168.11.123:5060 --->
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ACK sip:214019@192.168.11.124 SIP/2.0
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Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4
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CSeq: 7380 ACK
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From: <sip:214077@192.168.11.124>;tag=44166B53
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To: <sip:214019@192.168.11.124>;tag=as0353e318
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Call-ID: 1961359474@192.168.11.123
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Content-Length: 0
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User-Agent: kphone/4.2
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Contact: <sip:214077@192.168.11.123;transport=udp>
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<------------->
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--- (9 headers 0 lines) ---
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Really destroying SIP dialog '1961359474@192.168.11.123' Method: ACK
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bee*CLI> sip no debug
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SIP Debugging Disabled
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bee*CLI>