rendered paste bodysip debug peer 600
SIP Debugging Enabled for IP: 10.1.1.114:5060
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
TWAsterisk*CLI>
<--- SIP read from 10.1.1.114:5060 --->
INVITE sip:7167069363@10.1.1.202:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK28daca88886B56AB
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=6F8F873F-735336
To: <sip:7167069363@10.1.1.202;user=phone>
CSeq: 1 INVITE
Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114
Contact: <sip:600@10.1.1.114>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 257
v=0
o=- 1175878434 1175878434 IN IP4 10.1.1.114
s=Polycom IP Phone
c=IN IP4 10.1.1.114
t=0 0
m=audio 2244 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (14 headers 11 lines) ---
Sending to 10.1.1.114 : 5060 (no NAT)
Using INVITE request as basis request - 6cae04b2-13d7de4d-e790056c@10.1.1.114
<--- Reliably Transmitting (no NAT) to 10.1.1.114:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK28daca88886B56AB;received=10.1.1.114
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=6F8F873F-735336
To: <sip:7167069363@10.1.1.202;user=phone>;tag=as41904945
Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="32e9ec6c"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '6cae04b2-13d7de4d-e790056c@10.1.1.114' in 32000 ms (Method: INVITE)
Found user '600'
TWAsterisk*CLI>
<--- SIP read from 10.1.1.114:5060 --->
ACK sip:7167069363@10.1.1.202:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK28daca88886B56AB
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=6F8F873F-735336
To: <sip:7167069363@10.1.1.202;user=phone>;tag=as41904945
CSeq: 1 ACK
Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114
Contact: <sip:600@10.1.1.114>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127
Max-Forwards: 70
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from 10.1.1.114:5060 --->
INVITE sip:7167069363@10.1.1.202:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK853d060151BB7890
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=6F8F873F-735336
To: <sip:7167069363@10.1.1.202;user=phone>
CSeq: 2 INVITE
Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114
Contact: <sip:600@10.1.1.114>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="600", realm="asterisk", nonce="32e9ec6c", uri="sip:7167069363@10.1.1.202:5060;user=phone", response="08e1192f46bd636a36abf9e7ffed1c91", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 257
v=0
o=- 1175878434 1175878434 IN IP4 10.1.1.114
s=Polycom IP Phone
c=IN IP4 10.1.1.114
t=0 0
m=audio 2244 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (15 headers 11 lines) ---
Sending to 10.1.1.114 : 5060 (no NAT)
Using INVITE request as basis request - 6cae04b2-13d7de4d-e790056c@10.1.1.114
Found user '600'
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.1.1.114:2244
Found description format G722 for ID 9
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.114:2244
Looking for 7167069363 in default (domain 10.1.1.202)
<--- Reliably Transmitting (no NAT) to 10.1.1.114:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK853d060151BB7890;received=10.1.1.114
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=6F8F873F-735336
To: <sip:7167069363@10.1.1.202;user=phone>;tag=as41904945
Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '6cae04b2-13d7de4d-e790056c@10.1.1.114' in 32000 ms (Method: INVITE)
TWAsterisk*CLI>
<--- SIP read from 10.1.1.114:5060 --->
ACK sip:7167069363@10.1.1.202:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK853d060151BB7890
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=6F8F873F-735336
To: <sip:7167069363@10.1.1.202;user=phone>;tag=as41904945
CSeq: 2 ACK
Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114
Contact: <sip:600@10.1.1.114>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127
Proxy-Authorization: Digest username="600", realm="asterisk", nonce="32e9ec6c", uri="sip:7167069363@10.1.1.202:5060;user=phone", response="08e1192f46bd636a36abf9e7ffed1c91", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
TWAsterisk*CLI>
<--- SIP read from 10.1.1.114:5060 --->
REGISTER sip:10.1.1.202:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bKf1ffbafeA5B83129
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=59F81183-5EC2D3AA
To: <sip:600@10.1.1.202>
CSeq: 23 REGISTER
Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114
Contact: <sip:600@10.1.1.114>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127
Authorization: Digest username="600", realm="asterisk", nonce="1a751fb6", uri="sip:10.1.1.202:5060", response="d7a23b2cef79adc7e1a7bc0666bdf8e1", algorithm=MD5
Max-Forwards: 70
Expires: 30
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.114 : 5060 (no NAT)
<--- Transmitting (no NAT) to 10.1.1.114:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bKf1ffbafeA5B83129;received=10.1.1.114
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=59F81183-5EC2D3AA
To: <sip:600@10.1.1.202>
Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114
CSeq: 23 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:600@10.1.1.202>
Content-Length: 0
<------------>
TWAsterisk*CLI>
<--- Transmitting (no NAT) to 10.1.1.114:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bKf1ffbafeA5B83129;received=10.1.1.114
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=59F81183-5EC2D3AA
To: <sip:600@10.1.1.202>;tag=as01a16496
Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114
CSeq: 23 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:600@10.1.1.114>;expires=60
Date: Fri, 06 Apr 2007 17:08:07 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '19723fe6-fc84eef1-9931e0c0@10.1.1.114' in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog '1652f5625ef14199368414bb37a0a960@10.1.1.202' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 10.1.1.114:5060:
NOTIFY sip:600@10.1.1.114 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.202:5060;branch=z9hG4bK3c19d2cf;rport
From: "asterisk" <sip:asterisk@10.1.1.202>;tag=as4799aba1
To: <sip:600@10.1.1.114>
Contact: <sip:asterisk@10.1.1.202>
Call-ID: 1652f5625ef14199368414bb37a0a960@10.1.1.202
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 91
Messages-Waiting: yes
Message-Account: sip:asterisk@10.1.1.202
Voice-Message: 6/0 (0/0)
---
TWAsterisk*CLI>
<--- SIP read from 10.1.1.114:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.202:5060;branch=z9hG4bK3c19d2cf;rport
From: "asterisk" <sip:asterisk@10.1.1.202>;tag=as4799aba1
To: <sip:600@10.1.1.114>;tag=E25D798-87EFEC7B
CSeq: 102 NOTIFY
Call-ID: 1652f5625ef14199368414bb37a0a960@10.1.1.202
Contact: <sip:600@10.1.1.114>
Event: message-summary
User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1652f5625ef14199368414bb37a0a960@10.1.1.202' Method: NOTIFY
Really destroying SIP dialog '211e5122-efa4f3fd-d4d0c35c@10.1.1.114' Method: ACK
Really destroying SIP dialog '6cae04b2-13d7de4d-e790056c@10.1.1.114' Method: ACK
TWAsterisk*CLI>
<--- SIP read from 10.1.1.114:5060 --->
REGISTER sip:10.1.1.202:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK61e29c42891D5C9D
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=59F81183-5EC2D3AA
To: <sip:600@10.1.1.202>
CSeq: 24 REGISTER
Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114
Contact: <sip:600@10.1.1.114>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127
Authorization: Digest username="600", realm="asterisk", nonce="1a751fb6", uri="sip:10.1.1.202:5060", response="d7a23b2cef79adc7e1a7bc0666bdf8e1", algorithm=MD5
Max-Forwards: 70
Expires: 30
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.114 : 5060 (no NAT)
<--- Transmitting (no NAT) to 10.1.1.114:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK61e29c42891D5C9D;received=10.1.1.114
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=59F81183-5EC2D3AA
To: <sip:600@10.1.1.202>
Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114
CSeq: 24 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:600@10.1.1.202>
Content-Length: 0
<------------>
TWAsterisk*CLI>
<--- Transmitting (no NAT) to 10.1.1.114:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK61e29c42891D5C9D;received=10.1.1.114
From: "Chris_Martin" <sip:600@10.1.1.202>;tag=59F81183-5EC2D3AA
To: <sip:600@10.1.1.202>;tag=as01a16496
Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114
CSeq: 24 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:600@10.1.1.114>;expires=60
Date: Fri, 06 Apr 2007 17:08:37 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '19723fe6-fc84eef1-9931e0c0@10.1.1.114' in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog '32d41e722bb90fb06914018a56ab468b@10.1.1.202' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 10.1.1.114:5060:
NOTIFY sip:600@10.1.1.114 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.202:5060;branch=z9hG4bK7d3cb859;rport
From: "asterisk" <sip:asterisk@10.1.1.202>;tag=as143bf4c7
To: <sip:600@10.1.1.114>
Contact: <sip:asterisk@10.1.1.202>
Call-ID: 32d41e722bb90fb06914018a56ab468b@10.1.1.202
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 91
Messages-Waiting: yes
Message-Account: sip:asterisk@10.1.1.202
Voice-Message: 6/0 (0/0)
---
TWAsterisk*CLI>
<--- SIP read from 10.1.1.114:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.202:5060;branch=z9hG4bK7d3cb859;rport
From: "asterisk" <sip:asterisk@10.1.1.202>;tag=as143bf4c7
To: <sip:600@10.1.1.114>;tag=9497CB7C-1E36420F
CSeq: 102 NOTIFY
Call-ID: 32d41e722bb90fb06914018a56ab468b@10.1.1.202
Contact: <sip:600@10.1.1.114>
Event: message-summary
User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '32d41e722bb90fb06914018a56ab468b@10.1.1.202' Method: NOTIFY
TWAsterisk*CLI>