rendered paste bodyContent-Length: 0
(no NAT) to 10.0.0.7:5061
asterisk1*CLI>
Sip read:
SIP/2.0 200 OK
To: <sip:10.0.0.7:5061>;tag=e88b4295ac82497di1
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as24ac9e8b
Call-ID: 67796c8c2e1c54dc51f1450852d21d09@10.0.0.3
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK70dfbb01
Server: Sipura/SPA3000-3.1.5(GWb)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
10 headers, 0 lines
Destroying call '67796c8c2e1c54dc51f1450852d21d09@10.0.0.3'
asterisk1*CLI>
Sip read:
0 headers, 0 lines
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:201@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK75b94ad3
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as377dbb7c
To: <sip:201@10.0.0.1:5060>
Contact: <sip:Unknown@10.0.0.3>
Call-ID: 669ad9f470f04ae44486c11a1e2d9dd6@10.0.0.3
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 43
Messages-Waiting: yes
Voice-Message: 4/0
(no NAT) to 10.0.0.1:5060
Scheduling destruction of call '669ad9f470f04ae44486c11a1e2d9dd6@10.0.0.3' in 15000 ms
asterisk1*CLI>
Sip read:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK75b94ad3
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as377dbb7c
To: <sip:201@10.0.0.1:5060>;tag=2560275273
Contact: <sip:201@10.0.0.1:5060>
Call-ID: 669ad9f470f04ae44486c11a1e2d9dd6@10.0.0.3
CSeq: 102 NOTIFY
Server: X-Lite release 1103m
Content-Length: 0
9 headers, 0 lines
Destroying call '669ad9f470f04ae44486c11a1e2d9dd6@10.0.0.3'
Destroying call '77914300948549558338D039B8B5AA5D@10.0.0.3'
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.0.0.7:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5688ddb1
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as21b73c53
To: <sip:10.0.0.7:5061>
Contact: <sip:Unknown@10.0.0.3>
Call-ID: 1e765a327444aa9f4958567d3a517bcf@10.0.0.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Thu, 11 Aug 2005 18:46:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 10.0.0.7:5061
asterisk1*CLI>
Sip read:
SIP/2.0 200 OK
To: <sip:10.0.0.7:5061>;tag=e88b4295ac82497di1
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as21b73c53
Call-ID: 1e765a327444aa9f4958567d3a517bcf@10.0.0.3
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5688ddb1
Server: Sipura/SPA3000-3.1.5(GWb)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
10 headers, 0 lines
Destroying call '1e765a327444aa9f4958567d3a517bcf@10.0.0.3'
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.0.0.7:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK11300256
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as40d19cba
To: <sip:10.0.0.7:5061>
Contact: <sip:Unknown@10.0.0.3>
Call-ID: 5d664e917d9c08f86b53ff414be848f4@10.0.0.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Thu, 11 Aug 2005 18:47:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 10.0.0.7:5061
asterisk1*CLI>
Sip read:
SIP/2.0 200 OK
To: <sip:10.0.0.7:5061>;tag=e88b4295ac82497di1
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as40d19cba
Call-ID: 5d664e917d9c08f86b53ff414be848f4@10.0.0.3
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK11300256
Server: Sipura/SPA3000-3.1.5(GWb)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
10 headers, 0 lines
Destroying call '5d664e917d9c08f86b53ff414be848f4@10.0.0.3'
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
INVITE sip:201@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.7:5061;branch=z9hG4bK-e842bd88
From: PSTN <sip:38095871@10.0.0.3>;tag=be838a157c2fc500o1
To: <sip:201@10.0.0.3>
Call-ID: cc21b79-f321c2ec@10.0.0.7
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PSTN <sip:38095871@10.0.0.7:5061>
Expires: 240
User-Agent: Sipura/SPA3000-3.1.5(GWb)
Content-Length: 416
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 5578130 5578130 IN IP4 10.0.0.7
s=-
c=IN IP4 10.0.0.7
t=0 0
m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
14 headers, 19 lines
Using latest request as basis request
Sending to 10.0.0.7 : 5061 (non-NAT)
Found peer 'astrasip'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.7:16458
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 201 in from-pstn
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.7:5061;branch=z9hG4bK-e842bd88
From: PSTN <sip:38095871@10.0.0.3>;tag=be838a157c2fc500o1
To: <sip:201@10.0.0.3>;tag=as443c84e0
Call-ID: cc21b79-f321c2ec@10.0.0.7
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201@10.0.0.3>
Content-Length: 0
to 10.0.0.7:5061
asterisk1*CLI>
Sip read:
ACK sip:201@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.7:5061;branch=z9hG4bK-e842bd88
From: PSTN <sip:38095871@10.0.0.3>;tag=be838a157c2fc500o1
To: <sip:201@10.0.0.3>;tag=as443c84e0
Call-ID: cc21b79-f321c2ec@10.0.0.7
CSeq: 101 ACK
Max-Forwards: 70
Contact: PSTN <sip:38095871@10.0.0.7:5061>
User-Agent: Sipura/SPA3000-3.1.5(GWb)
Content-Length: 0
10 headers, 0 lines
Destroying call 'cc21b79-f321c2ec@10.0.0.7'
asterisk1*CLI>
Sip read:
INVITE sip:201@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.7:5061;branch=z9hG4bK-4aedf047
From: PSTN <sip:PSTN@10.0.0.3>;tag=d86bb82f2022e007o1
To: <sip:201@10.0.0.3>
Call-ID: 9d83994f-bf77167@10.0.0.7
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PSTN <sip:PSTN@10.0.0.7:5061>
Expires: 240
User-Agent: Sipura/SPA3000-3.1.5(GWb)
Content-Length: 416
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 5578731 5578731 IN IP4 10.0.0.7
s=-
c=IN IP4 10.0.0.7
t=0 0
m=audio 16464 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
14 headers, 19 lines
Using latest request as basis request
Sending to 10.0.0.7 : 5061 (non-NAT)
Found peer 'astrasip'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.7:16464
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 201 in from-pstn
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.7:5061;branch=z9hG4bK-4aedf047
From: PSTN <sip:PSTN@10.0.0.3>;tag=d86bb82f2022e007o1
To: <sip:201@10.0.0.3>;tag=as46c5c582
Call-ID: 9d83994f-bf77167@10.0.0.7
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201@10.0.0.3>
Content-Length: 0
to 10.0.0.7:5061
asterisk1*CLI>
Sip read:
ACK sip:201@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.7:5061;branch=z9hG4bK-4aedf047
From: PSTN <sip:PSTN@10.0.0.3>;tag=d86bb82f2022e007o1
To: <sip:201@10.0.0.3>;tag=as46c5c582
Call-ID: 9d83994f-bf77167@10.0.0.7
CSeq: 101 ACK
Max-Forwards: 70
Contact: PSTN <sip:PSTN@10.0.0.7:5061>
User-Agent: Sipura/SPA3000-3.1.5(GWb)
Content-Length: 0
10 headers, 0 lines
Destroying call '9d83994f-bf77167@10.0.0.7'
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.0.0.7:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK2f82b3db
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as143eaac3
To: <sip:10.0.0.7:5061>
Contact: <sip:Unknown@10.0.0.3>
Call-ID: 138b81e600ba36d66ca1f91662648e15@10.0.0.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Thu, 11 Aug 2005 18:48:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 10.0.0.7:5061
asterisk1*CLI>
Sip read:
SIP/2.0 200 OK
To: <sip:10.0.0.7:5061>;tag=e88b4295ac82497di1
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as143eaac3
Call-ID: 138b81e600ba36d66ca1f91662648e15@10.0.0.3
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK2f82b3db
Server: Sipura/SPA3000-3.1.5(GWb)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
10 headers, 0 lines
Destroying call '138b81e600ba36d66ca1f91662648e15@10.0.0.3'
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.0.0.7:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK31f80ce4
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as540f5501
To: <sip:10.0.0.7:5061>
Contact: <sip:Unknown@10.0.0.3>
Call-ID: 07644d4b5a26dd0624b9abd1341e4194@10.0.0.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Thu, 11 Aug 2005 18:49:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 10.0.0.7:5061
asterisk1*CLI>
Sip read:
SIP/2.0 200 OK
To: <sip:10.0.0.7:5061>;tag=e88b4295ac82497di1
From: "Unknown" <sip:Unknown@10.0.0.3>;tag=as540f5501
Call-ID: 07644d4b5a26dd0624b9abd1341e4194@10.0.0.3
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK31f80ce4
Server: Sipura/SPA3000-3.1.5(GWb)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
10 headers, 0 lines
Destroying call '07644d4b5a26dd0624b9abd1341e4194@10.0.0.3'
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>
Sip read:
0 headers, 0 lines
asterisk1*CLI>