Public feed #381806

Znuff

public text v1 · immutable
Raw Edit
·published 2007-03-05 02:33 UTC
rendered paste body
From: "23002" <sip:23002@voip.telefonet.ro>;tag=aabpsKuK5JPcWtSE
To: "23002" <sip:23002@voip.telefonet.ro>
Call-ID: owiz95Hz9XJxvP8b@89.35.79.94
CSeq: 12407 REGISTER
Contact: <sip:23002@89.35.79.94:5060>
Expires: 60
Authorization: Digest username="23002", realm="asterisk", nonce="3b055b23", uri="sip:voip.telefonet.ro", response="e40b04325817245090db2d9d919f84c8"
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 89.35.79.94 : 5060 (no NAT)

<--- Transmitting (no NAT) to 89.35.79.94:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.35.79.94:5060;branch=z9hG4bKIpZoJszot;received=89.35.79.94
From: "23002" <sip:23002@voip.telefonet.ro>;tag=aabpsKuK5JPcWtSE
To: "23002" <sip:23002@voip.telefonet.ro>
Call-ID: owiz95Hz9XJxvP8b@89.35.79.94
CSeq: 12407 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23002@89.35.79.5>
Content-Length: 0


<------------>
voip*CLI>
<--- Transmitting (no NAT) to 89.35.79.94:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.94:5060;branch=z9hG4bKIpZoJszot;received=89.35.79.94
From: "23002" <sip:23002@voip.telefonet.ro>;tag=aabpsKuK5JPcWtSE
To: "23002" <sip:23002@voip.telefonet.ro>;tag=as26ffa75b
Call-ID: owiz95Hz9XJxvP8b@89.35.79.94
CSeq: 12407 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:23002@89.35.79.94:5060>;expires=60
Date: Mon, 05 Mar 2007 02:27:24 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'owiz95Hz9XJxvP8b@89.35.79.94' in 32000 ms (Method: REGISTER)
voip*CLI>
Scheduling destruction of SIP dialog '1970aa3b1b4b2cb87544201012352ae2@89.35.79.5' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 89.35.79.247:5060:
NOTIFY sip:23001@89.35.79.247:5060 SIP/2.0
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK1be45660;rport
From: "asterisk" <sip:asterisk@89.35.79.5>;tag=as48689ba2
To: <sip:23001@89.35.79.247:5060>
Contact: <sip:asterisk@89.35.79.5>
Call-ID: 1970aa3b1b4b2cb87544201012352ae2@89.35.79.5
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@89.35.79.5
Voice-Message: 1/0 (0/0)

---
voip*CLI>
<--- SIP read from 89.35.79.247:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK1be45660;rport
Call-ID: 1970aa3b1b4b2cb87544201012352ae2@89.35.79.5
CSeq: 102 NOTIFY
From: "asterisk" <sip:asterisk@89.35.79.5>;tag=as48689ba2
To: <sip:23001@89.35.79.247:5060>;tag=HXL4XOOmb9oG7f9l
Contact: <sip:23001@89.35.79.247:5060>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1970aa3b1b4b2cb87544201012352ae2@89.35.79.5' Method: NOTIFY
Scheduling destruction of SIP dialog '588ae2d34868bd283eeb09b42cc02107@89.35.79.5' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 89.35.79.94:5060:
NOTIFY sip:23002@89.35.79.94:5060 SIP/2.0
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK13d2c8e6;rport
From: "asterisk" <sip:asterisk@89.35.79.5>;tag=as3c1621ae
To: <sip:23002@89.35.79.94:5060>
Contact: <sip:asterisk@89.35.79.5>
Call-ID: 588ae2d34868bd283eeb09b42cc02107@89.35.79.5
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 90

Messages-Waiting: no
Message-Account: sip:asterisk@89.35.79.5
Voice-Message: 0/0 (0/0)

---
voip*CLI>
<--- SIP read from 89.35.79.94:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK13d2c8e6;rport
Call-ID: 588ae2d34868bd283eeb09b42cc02107@89.35.79.5
CSeq: 102 NOTIFY
From: "asterisk" <sip:asterisk@89.35.79.5>;tag=as3c1621ae
To: <sip:23002@89.35.79.94:5060>;tag=Hs3pN1qXMhUlF7VQ
Contact: <sip:23002@89.35.79.94:5060>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '588ae2d34868bd283eeb09b42cc02107@89.35.79.5' Method: NOTIFY
voip*CLI>
<--- SIP read from 89.35.79.247:5060 --->
INVITE sip:23003@voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKW227kVZvZ
Max-Forwards: 70
User-Agent: Wu Chuan
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YG6UP1oYxeS4ilVp
To: "23003" <sip:23003@voip.telefonet.ro>
Call-ID: SPbbitaAMsMutV8Q@89.35.79.247
Contact: <sip:23001@89.35.79.247:5060>
CSeq: 1 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE
Supported: 100rel, replaces
Content-Type: application/sdp
Content-Length: 215

v=0
o=23001 23013746 72060732 IN IP4 89.35.79.247
s=SIP CALL
c=IN IP4 89.35.79.247
t=0 0
m=audio 30000 RTP/AVP 18 4 0 8
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

<------------->
--- (13 headers 10 lines) ---
Sending to 89.35.79.247 : 5060 (no NAT)
Using INVITE request as basis request - SPbbitaAMsMutV8Q@89.35.79.247
Found peer '23001'

<--- Reliably Transmitting (no NAT) to 89.35.79.247:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKW227kVZvZ;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YG6UP1oYxeS4ilVp
To: "23003" <sip:23003@voip.telefonet.ro>;tag=as5ad4b123
Call-ID: SPbbitaAMsMutV8Q@89.35.79.247
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="25942128"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'SPbbitaAMsMutV8Q@89.35.79.247' in 32000 ms (Method: INVITE)
voip*CLI>
<--- SIP read from 89.35.79.247:5060 --->
ACK sip:23003@voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKW227kVZvZ
Max-Forwards: 70
User-Agent: Wu Chuan
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YG6UP1oYxeS4ilVp
To: "23003" <sip:23003@voip.telefonet.ro>;tag=as5ad4b123
Call-ID: SPbbitaAMsMutV8Q@89.35.79.247
Contact: <sip:23001@89.35.79.247:5060>
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
voip*CLI>
<--- SIP read from 89.35.79.247:5060 --->
INVITE sip:23003@voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv
Max-Forwards: 70
User-Agent: Wu Chuan
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YG6UP1oYxeS4ilVp
To: "23003" <sip:23003@voip.telefonet.ro>
Call-ID: SPbbitaAMsMutV8Q@89.35.79.247
Contact: <sip:23001@89.35.79.247:5060>
Proxy-Authorization: Digest username="23001", realm="asterisk", nonce="25942128", uri="sip:23003@voip.telefonet.ro", response="b12da7f6a02fc62a45fe9453d928853d"
CSeq: 2 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE
Supported: 100rel, replaces
Content-Type: application/sdp
Content-Length: 215

v=0
o=23001 67637671 39620179 IN IP4 89.35.79.247
s=SIP CALL
c=IN IP4 89.35.79.247
t=0 0
m=audio 30000 RTP/AVP 18 4 0 8
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

<------------->
--- (14 headers 10 lines) ---
Sending to 89.35.79.247 : 5060 (no NAT)
Using INVITE request as basis request - SPbbitaAMsMutV8Q@89.35.79.247
Found peer '23001'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port 89.35.79.247:30000
Found description format G729 for ID 18
Found description format G723 for ID 4
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 89.35.79.247:30000
Looking for 23003 in numberplan-custom-1 (domain voip.telefonet.ro)
list_route: hop: <sip:23001@89.35.79.247:5060>

<--- Transmitting (no NAT) to 89.35.79.247:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YG6UP1oYxeS4ilVp
To: "23003" <sip:23003@voip.telefonet.ro>
Call-ID: SPbbitaAMsMutV8Q@89.35.79.247
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@89.35.79.5>
Content-Length: 0


<------------>
    -- Executing [23003@numberplan-custom-1:1] Macro("SIP/23001-b5b0f3d8", "trunkdial|SIP/trunk_1/23003") in new stack
    -- Executing [s@macro-trunkdial:1] Dial("SIP/23001-b5b0f3d8", "SIP/trunk_1/23003") in new stack
Audio is at 10.0.0.2 port 19902
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.1:5060:
INVITE sip:23003@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK05b1673f;rport
From: "Test" <sip:23001@10.0.0.2>;tag=as67d70ee4
To: <sip:23003@10.0.0.1>
Contact: <sip:23001@10.0.0.2>
Call-ID: 47b5de4a4d5113581f9bc6c81d2becb7@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 05 Mar 2007 02:27:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6308 6308 IN IP4 89.35.79.247
s=session
c=IN IP4 89.35.79.247
t=0 0
m=audio 30000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called trunk_1/23003
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 100 Trying
From: "Test"<sip:23001@10.0.0.2>;tag=as67d70ee4
To: <sip:23003@10.0.0.1>;tag=a000001-13c4-96f-24f9c8-a0f
Call-ID: 47b5de4a4d5113581f9bc6c81d2becb7@10.0.0.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK05b1673f;rport
Contact: <sip:23003@10.0.0.1>
Content-Length:0


<------------->
--- (8 headers 0 lines) ---
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
INVITE sip:23003@10.0.0.2;transport=UDP;TCID-27 SIP/2.0
From: "TEST"<sip:23001@10.0.0.1:5060>;tag=a000001-13c4-970-24fb54-382b
To: <sip:23003@10.0.0.2>
Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488
Max-Forwards: 70
Contact: <sip:23001@10.0.0.1:5060;transport=UDP>
Content-Type: application/SDP
Content-Length:288

v=0
o=a0027 1234567890 1234567890 IN IP4 10.0.0.1
s=SIP Call test
c=IN IP4 10.0.0.1
t=0 0
m=audio 10000 RTP/AVP 8 97 0 18
a=fmtp:97 0-15
a=fmtp:18 annexb=yes
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000

<------------->
--- (10 headers 13 lines) ---
Sending to 10.0.0.1 : 5060 (no NAT)
Using INVITE request as basis request - 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1
Found peer 'trunk_1'
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 10.0.0.1:10000
Found description format PCMA for ID 8
Found description format telephone-event for ID 97
Found description format PCMU for ID 0
Found description format G729 for ID 18
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/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 10.0.0.1:10000
Looking for 23003 in DID_trunk_1 (domain 10.0.0.2)
list_route: hop: <sip:23001@10.0.0.1:5060;transport=UDP>
voip*CLI>
<--- Transmitting (no NAT) to 10.0.0.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488;received=10.0.0.1
From: "TEST"<sip:23001@10.0.0.1:5060>;tag=a000001-13c4-970-24fb54-382b
To: <sip:23003@10.0.0.2>
Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Length: 0


<------------>
    -- Executing [23003@DID_trunk_1:1] Dial("SIP/5060-b5b13340", "SIP/23003") in new stack
Audio is at 89.35.79.5 port 16240
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 89.35.79.69:39246:
INVITE sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b SIP/2.0
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport
From: "TEST" <sip:23001@89.35.79.5>;tag=as46e6ae2d
To: <sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b>
Contact: <sip:23001@89.35.79.5>
Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 05 Mar 2007 02:27:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 6308 6308 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 10000 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 23003
voip*CLI>
<--- SIP read from 89.35.79.69:39246 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport=5060
Contact: <sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b>
To: <sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b>;tag=b9473829
From: "TEST"<sip:23001@89.35.79.5>;tag=as46e6ae2d
Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5
CSeq: 102 INVITE
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/23003-08277f78 is ringing

<--- Transmitting (no NAT) to 10.0.0.1:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488;received=10.0.0.1
From: "TEST"<sip:23001@10.0.0.1:5060>;tag=a000001-13c4-970-24fb54-382b
To: <sip:23003@10.0.0.2>;tag=as788a2f98
Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Length: 0


<------------>
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 415 Unsupported Media Type
From: "Test"<sip:23001@10.0.0.2>;tag=as67d70ee4
To: <sip:23003@10.0.0.1>;tag=a000001-13c4-96f-24f9c8-a0f
Call-ID: 47b5de4a4d5113581f9bc6c81d2becb7@10.0.0.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK05b1673f;rport
Content-Length:0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 415 "Unsupported Media Type" back from 10.0.0.1
Transmitting (no NAT) to 10.0.0.1:5060:
ACK sip:23003@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK05b1673f;rport
From: "Test" <sip:23001@10.0.0.2>;tag=as67d70ee4
To: <sip:23003@10.0.0.1>;tag=a000001-13c4-96f-24f9c8-a0f
Contact: <sip:23001@10.0.0.2>
Call-ID: 47b5de4a4d5113581f9bc6c81d2becb7@10.0.0.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-trunkdial:2] Goto("SIP/23001-b5b0f3d8", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-trunkdial,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-trunkdial:1] NoOp("SIP/23001-b5b0f3d8", "") in new stack
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
CANCEL sip:23003@10.0.0.2;transport=UDP;TCID-27 SIP/2.0
From: <sip:23001@10.0.0.1:5060>;tag=a000001-13c4-970-24fb54-382b
To: <sip:23003@10.0.0.2>
Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1
CSeq: 1 CANCEL
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488
Max-Forwards: 70
Content-Length:0


<------------->
--- (8 headers 0 lines) ---
Sending to 10.0.0.1 : 5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 10.0.0.1:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488;received=10.0.0.1
From: "TEST"<sip:23001@10.0.0.1:5060>;tag=a000001-13c4-970-24fb54-382b
To: <sip:23003@10.0.0.2>;tag=as788a2f98
Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


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

<--- Transmitting (no NAT) to 10.0.0.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488;received=10.0.0.1
From: <sip:23001@10.0.0.1:5060>;tag=a000001-13c4-970-24fb54-382b
To: <sip:23003@10.0.0.2>;tag=as788a2f98
Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1
CSeq: 1 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Length: 0


<------------>
Really destroying SIP dialog '47b5de4a4d5113581f9bc6c81d2becb7@10.0.0.2' Method: INVITE
Scheduling destruction of SIP dialog '49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 89.35.79.69:39246:
CANCEL sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b SIP/2.0
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport
From: "TEST" <sip:23001@89.35.79.5>;tag=as46e6ae2d
To: <sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b>
Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5' in 32000 ms (Method: INVITE)
  == Spawn extension (DID_trunk_1, 23003, 1) exited non-zero on 'SIP/5060-b5b13340'
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
ACK sip:23003@10.0.0.2;transport=UDP;TCID-27 SIP/2.0
From: <sip:23001@10.0.0.1:5060>;tag=a000001-13c4-970-24fb54-382b
To: <sip:23003@10.0.0.2>;tag=as788a2f98
Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488
Max-Forwards: 70
Contact: <sip:23001@10.0.0.1:5060;transport=UDP>
Content-Length:0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1' Method: ACK
voip*CLI>
<--- SIP read from 89.35.79.69:39246 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport=5060
Contact: <sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b>
To: <sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b>;tag=b9473829
From: "TEST"<sip:23001@89.35.79.5>;tag=as46e6ae2d
Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5
CSeq: 102 CANCEL
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
voip*CLI>
<--- SIP read from 89.35.79.69:39246 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport=5060
To: <sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b>;tag=b9473829
From: "TEST"<sip:23001@89.35.79.5>;tag=as46e6ae2d
Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5
CSeq: 102 INVITE
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 89.35.79.69:39246:
ACK sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b SIP/2.0
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK6a3c2910;rport
From: "TEST" <sip:23001@89.35.79.5>;tag=as46e6ae2d
To: <sip:23003@89.35.79.69:39246;rinstance=a9cce79c2a37b77b>;tag=b9473829
Contact: <sip:23001@89.35.79.5>
Call-ID: 49eac613368d6f0e1fc464a63cd65ee4@89.35.79.5
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
voip*CLI>
<--- SIP read from 89.35.79.247:5060 --->
CANCEL sip:23003@voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv
Max-Forwards: 70
User-Agent: Wu Chuan
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YG6UP1oYxeS4ilVp
To: "23003" <sip:23003@voip.telefonet.ro>
Call-ID: SPbbitaAMsMutV8Q@89.35.79.247
Contact: <sip:23001@89.35.79.247:5060>
Proxy-Authorization: Digest username="23001", realm="asterisk", nonce="25942128", uri="sip:23003@voip.telefonet.ro", response="b12da7f6a02fc62a45fe9453d928853d"
CSeq: 2 CANCEL
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 89.35.79.247 : 5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 89.35.79.247:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YG6UP1oYxeS4ilVp
To: "23003" <sip:23003@voip.telefonet.ro>;tag=as5ba87637
Call-ID: SPbbitaAMsMutV8Q@89.35.79.247
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


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

<--- Transmitting (no NAT) to 89.35.79.247:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YG6UP1oYxeS4ilVp
To: "23003" <sip:23003@voip.telefonet.ro>;tag=as5ba87637
Call-ID: SPbbitaAMsMutV8Q@89.35.79.247
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@89.35.79.5>
Content-Length: 0


<------------>
voip*CLI>
<--- SIP read from 89.35.79.247:5060 --->
ACK sip:23003@voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKfS4gesXYv
Max-Forwards: 70
User-Agent: Wu Chuan
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YG6UP1oYxeS4ilVp
To: "23003" <sip:23003@voip.telefonet.ro>
Call-ID: SPbbitaAMsMutV8Q@89.35.79.247
Contact: <sip:23001@89.35.79.247:5060>
Proxy-Authorization: Digest username="23001", realm="asterisk", nonce="25942128", uri="sip:23003@voip.telefonet.ro", response="b12da7f6a02fc62a45fe9453d928853d"
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog 'SPbbitaAMsMutV8Q@89.35.79.247' Method: ACK
voip*CLI>