All pastes #557438 Raw Edit

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public text v1 · immutable
·published 2007-06-11 00:01 UTC
rendered paste body
asterisk*CLI>
<-- SIP read from 64.34.176.212:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.205:5060;branch=z9hG4bK49d1100c;received=68.149.47.230;rport=5060
From: <sip:1213214177@did.voip.les.net>;tag=as7d2ab5d0
To: <sip:1213214177@did.voip.les.net>
Call-ID: 30e04fa67db52261382a61a6120cb247@192.168.1.205
CSeq: 125 REGISTER
User-Agent: LES.NET.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1213214177@64.34.176.212>
Content-Length: 0


--- (10 headers 0 lines) ---
asterisk*CLI>
<-- SIP read from 64.34.176.212:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.205:5060;branch=z9hG4bK49d1100c;received=68.149.47.230;rport=5060
From: <sip:1213214177@did.voip.les.net>;tag=as7d2ab5d0
To: <sip:1213214177@did.voip.les.net>;tag=as40f3e2c3
Call-ID: 30e04fa67db52261382a61a6120cb247@192.168.1.205
CSeq: 125 REGISTER
User-Agent: LES.NET.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 120
Contact: <sip:1213214177@192.168.1.205>;expires=120
Date: Mon, 11 Jun 2007 00:00:16 GMT
Content-Length: 0


--- (12 headers 0 lines) ---
Scheduling destruction of call '30e04fa67db52261382a61a6120cb247@192.168.1.205' in 32000 ms
asterisk*CLI>
<-- SIP read from 64.34.176.212:5060:
INVITE sip:17806283501@68.149.47.230 SIP/2.0
Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK0186bb4d;rport
From: "7807097214" <sip:7807097214@64.34.176.212>;tag=as52c703ef
To: <sip:17806283501@68.149.47.230>
Contact: <sip:7807097214@64.34.176.212>
Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
CSeq: 102 INVITE
User-Agent: LES.NET.VoIP
Max-Forwards: 70
Date: Mon, 11 Jun 2007 00:00:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 6451 6451 IN IP4 64.34.176.212
s=session
c=IN IP4 64.34.176.212
t=0 0
m=audio 10478 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (13 headers 10 lines) ---
Using INVITE request as basis request - 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
Sending to 64.34.176.212 : 5060 (NAT)
Found peer '1213214177'
Reliably Transmitting (no NAT) to 64.34.176.212:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK0186bb4d;rport;received=64.34.176.212
From: "7807097214" <sip:7807097214@64.34.176.212>;tag=as52c703ef
To: <sip:17806283501@68.149.47.230>;tag=as68ca19d6
Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="18850f89"
Content-Length: 0


---
Scheduling destruction of call '0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212' in 15000 ms
asterisk*CLI>
<-- SIP read from 64.34.176.212:5060:
ACK sip:17806283501@68.149.47.230 SIP/2.0
Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK0186bb4d;rport
From: "7807097214" <sip:7807097214@64.34.176.212>;tag=as52c703ef
To: <sip:17806283501@68.149.47.230>;tag=as68ca19d6
Contact: <sip:7807097214@64.34.176.212>
Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
CSeq: 102 ACK
User-Agent: LES.NET.VoIP
Max-Forwards: 70
Content-Length: 0


--- (10 headers 0 lines) ---
asterisk*CLI>
<-- SIP read from 64.34.176.212:5060:
INVITE sip:17806283501@68.149.47.230 SIP/2.0
Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK40090ebf;rport
From: "7807097214" <sip:7807097214@64.34.176.212>;tag=as52c703ef
To: <sip:17806283501@68.149.47.230>
Contact: <sip:7807097214@64.34.176.212>
Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
CSeq: 103 INVITE
User-Agent: LES.NET.VoIP
Max-Forwards: 70
Proxy-Authorization: Digest username="1213214177", realm="asterisk", algorithm=MD5, uri="sip:17806283501@68.149.47.230", nonce="18850f89", response="9c42c95c021697f7aaeec069ef2e818f", opaque=""
Date: Mon, 11 Jun 2007 00:00:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 6451 6452 IN IP4 64.34.176.212
s=session
c=IN IP4 64.34.176.212
t=0 0
m=audio 10478 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (14 headers 10 lines) ---
Using INVITE request as basis request - 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
Sending to 64.34.176.212 : 5060 (NAT)
Found peer '1213214177'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 64.34.176.212:10478
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 17806283501 in from-sip-external (domain 68.149.47.230)
list_route: hop: <sip:7807097214@64.34.176.212>
Transmitting (no NAT) to 64.34.176.212:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK40090ebf;rport;received=64.34.176.212
From: "7807097214" <sip:7807097214@64.34.176.212>;tag=as52c703ef
To: <sip:17806283501@68.149.47.230>
Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:17806283501@192.168.1.205>
Content-Length: 0


---
    -- Executing NoOp("SIP/1213214177-0818b020", "Received incoming SIP connection from unknown peer to 17806283501") in new stack
    -- Executing Set("SIP/1213214177-0818b020", "DID=17806283501") in new stack
    -- Executing Goto("SIP/1213214177-0818b020", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing Ringing("SIP/1213214177-0818b020", "") in new stack
Transmitting (no NAT) to 64.34.176.212:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK40090ebf;rport;received=64.34.176.212
From: "7807097214" <sip:7807097214@64.34.176.212>;tag=as52c703ef
To: <sip:17806283501@68.149.47.230>;tag=as2e4fe317
Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:17806283501@192.168.1.205>
Content-Length: 0


---
    -- Executing GotoIf("SIP/1213214177-0818b020", "0?from-trunk|17806283501|1") in new stack
    -- Executing Set("SIP/1213214177-0818b020", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2007-06-11 00:04:56 UTC.
    -- Executing Answer("SIP/1213214177-0818b020", "") in new stack
We're at 192.168.1.205 port 13810
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.34.176.212:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK40090ebf;rport;received=64.34.176.212
From: "7807097214" <sip:7807097214@64.34.176.212>;tag=as52c703ef
To: <sip:17806283501@68.149.47.230>;tag=as2e4fe317
Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:17806283501@192.168.1.205>
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 3331 3331 IN IP4 192.168.1.205
s=session
c=IN IP4 192.168.1.205
t=0 0
m=audio 13810 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Executing Wait("SIP/1213214177-0818b020", "2") in new stack
asterisk*CLI>
<-- SIP read from 64.34.176.212:5060:
ACK sip:17806283501@192.168.1.205 SIP/2.0
Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK341877a6;rport
From: "7807097214" <sip:7807097214@64.34.176.212>;tag=as52c703ef
To: <sip:17806283501@68.149.47.230>;tag=as2e4fe317
Contact: <sip:7807097214@64.34.176.212>
Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
CSeq: 103 ACK
User-Agent: LES.NET.VoIP
Max-Forwards: 70
Content-Length: 0


--- (10 headers 0 lines) ---
    -- Executing Playback("SIP/1213214177-0818b020", "ss-noservice") in new stack
    -- Playing 'ss-noservice' (language 'en')
asterisk*CLI>
<-- SIP read from 64.34.176.212:5060:
BYE sip:17806283501@192.168.1.205 SIP/2.0
Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK591e2ac2;rport
From: "7807097214" <sip:7807097214@64.34.176.212>;tag=as52c703ef
To: <sip:17806283501@68.149.47.230>;tag=as2e4fe317
Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
CSeq: 104 BYE
User-Agent: LES.NET.VoIP
Max-Forwards: 70
Proxy-Authorization: Digest username="1213214177", realm="asterisk", algorithm=MD5, uri="sip:17806283501@192.168.1.205", nonce="18850f89", response="835da58da3de3aa5a733d1dbe3ac7f43", opaque=""
Content-Length: 0


--- (10 headers 0 lines) ---
Sending to 64.34.176.212 : 5060 (NAT)
Transmitting (NAT) to 64.34.176.212:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.34.176.212:5060;branch=z9hG4bK591e2ac2;received=64.34.176.212;rport=5060
From: "7807097214" <sip:7807097214@64.34.176.212>;tag=as52c703ef
To: <sip:17806283501@68.149.47.230>;tag=as2e4fe317
Call-ID: 0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212
CSeq: 104 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:17806283501@192.168.1.205>
Content-Length: 0


---
  == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/1213214177-0818b020'
    -- Executing NoOp("SIP/1213214177-0818b020", "Hangup") in new stack
    -- Executing Set("SIP/1213214177-0818b020", "DID=s") in new stack
    -- Executing Goto("SIP/1213214177-0818b020", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing Ringing("SIP/1213214177-0818b020", "") in new stack
    -- Executing GotoIf("SIP/1213214177-0818b020", "0?from-trunk|s|1") in new stack
    -- Executing Set("SIP/1213214177-0818b020", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2007-06-11 00:05:01 UTC.
    -- Executing Answer("SIP/1213214177-0818b020", "") in new stack
  == Spawn extension (from-sip-external, s, 4) exited non-zero on 'SIP/1213214177-0818b020'
Destroying call '0133d3a873b2da4239cc1f8641af4d0c@64.34.176.212'
Destroying call '30e04fa67db52261382a61a6120cb247@192.168.1.205'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.198:5060:
OPTIONS sip:2000@192.168.1.198:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.205:5060;branch=z9hG4bK19d9c663;rport
From: "Unknown" <sip:Unknown@192.168.1.205>;tag=as434bc03c
To: <sip:2000@192.168.1.198:5060>
Contact: <sip:Unknown@192.168.1.205>
Call-ID: 70ff071e37ab3ba349877c5f711bd7e4@192.168.1.205
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 11 Jun 2007 00:05:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from 192.168.1.198:5060:
SIP/2.0 200 OK
To: <sip:2000@192.168.1.198:5060>;tag=1b057364308db77ei0
From: "Unknown" <sip:Unknown@192.168.1.205>;tag=as434bc03c
Call-ID: 70ff071e37ab3ba349877c5f711bd7e4@192.168.1.205
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.205:5060;branch=z9hG4bK19d9c663;rport=5060
Server: Linksys/PAP2-3.1.12(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


--- (10 headers 0 lines) ---
Destroying call '70ff071e37ab3ba349877c5f711bd7e4@192.168.1.205'