All pastes #740078 Raw Edit

Anonymous

public text v1 · immutable
·published 2007-10-17 18:07 UTC
rendered paste body
asterisk1*CLI> 
<-- SIP read from 192.168.20.19:5060: 
INVITE sip:user_ip@192.168.20.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.19:5060;rport;branch=z9hG4bK08cf18272b
From: "5922261697" <sip:19@192.168.20.4>;tag=4b6f379d
To: <sip:user_ip@192.168.20.4>
Call-ID: 5ac46a11150fb0a539162b705e7de357@192.168.20.19
Contact: <sip:19@192.168.20.19:5060>
CSeq: 801 INVITE
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE
Supported: replaces
Content-Type: application/sdp
User-Agent: CM5K  (706220)
Content-Length: 393

v=0
o=CMI-SIPUA 38811 0 IN IP4 192.168.20.19
s=SIP CALL
c=IN IP4 192.168.20.19
t=0 0
m=audio 60000 RTP/AVP 0 8 18 4 23 22 2 21 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:23 G726-16/8000
a=rtpmap:22 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:21 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (13 headers 17 lines) ---
Using INVITE request as basis request - 5ac46a11150fb0a539162b705e7de357@192.168.20.19
Sending to 192.168.20.19 : 5060 (NAT)
Reliably Transmitting (no NAT) to 192.168.20.19:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.20.19:5060;branch=z9hG4bK08cf18272b;received=192.168.20.19;rport=5060
From: "5922261697" <sip:19@192.168.20.4>;tag=4b6f379d
To: <sip:user_ip@192.168.20.4>;tag=as607087e1
Call-ID: 5ac46a11150fb0a539162b705e7de357@192.168.20.19
CSeq: 801 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="074a38f0"
Content-Length: 0


---
Scheduling destruction of call '5ac46a11150fb0a539162b705e7de357@192.168.20.19' in 15000 ms
Found user '19'
asterisk1*CLI> 
<-- SIP read from 192.168.20.19:5060: 
ACK sip:user_ip@192.168.20.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.19:5060;branch=z9hG4bK08cf18272b
From: "5922261697" <sip:19@192.168.20.4>;tag=4b6f379d
To: <sip:user_ip@192.168.20.4>;tag=as607087e1
Call-ID: 5ac46a11150fb0a539162b705e7de357@192.168.20.19
Contact: <sip:19@192.168.20.19:5060>
CSeq: 801 ACK
Max-Forwards: 70
User-Agent: CM5K  (706220)
Content-Length: 0


--- (10 headers 0 lines) ---
asterisk1*CLI> 
<-- SIP read from 192.168.20.19:5060: 
INVITE sip:user_ip@192.168.20.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.19:5060;rport;branch=z9hG4bK31feac7e83
From: "5922261697" <sip:19@192.168.20.4>;tag=4b6f379d
To: <sip:user_ip@192.168.20.4>
Call-ID: 5ac46a11150fb0a539162b705e7de357@192.168.20.19
Contact: <sip:19@192.168.20.19:5060>
CSeq: 802 INVITE
Proxy-Authorization: Digest username="19",realm="asterisk",nonce="074a38f0",response="791edfd995d2e1294bfabae96b5434d0",uri="sip:user_ip@192.168.20.4",algorithm=MD5
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE
Supported: replaces
Content-Type: application/sdp
User-Agent: CM5K  (706220)
Content-Length: 393

v=0
o=CMI-SIPUA 38811 0 IN IP4 192.168.20.19
s=SIP CALL
c=IN IP4 192.168.20.19
t=0 0
m=audio 60000 RTP/AVP 0 8 18 4 23 22 2 21 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:23 G726-16/8000
a=rtpmap:22 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:21 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (14 headers 17 lines) ---
Using INVITE request as basis request - 5ac46a11150fb0a539162b705e7de357@192.168.20.19
Sending to 192.168.20.19 : 5060 (NAT)
Found user '19'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 23
Found RTP audio format 22
Found RTP audio format 2
Found RTP audio format 21
Found RTP audio format 101
Peer audio RTP is at port 192.168.20.19:60000
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G723
Found description format G726-16
Found description format G726-24
Found description format G726-32
Found description format G726-40
Found description format telephone-event
Capabilities: us - 0x50f (g723|gsm|ulaw|alaw|g729|ilbc), peer - audio=0x11d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x10d (g723|ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for user_ip in from-internal (domain 192.168.20.4)
Reliably Transmitting (no NAT) to 192.168.20.19:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.20.19:5060;branch=z9hG4bK31feac7e83;received=192.168.20.19;rport=5060
From: "5922261697" <sip:19@192.168.20.4>;tag=4b6f379d
To: <sip:user_ip@192.168.20.4>;tag=as607087e1
Call-ID: 5ac46a11150fb0a539162b705e7de357@192.168.20.19
CSeq: 802 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk1*CLI> 
<-- SIP read from 192.168.20.19:5060: 
ACK sip:user_ip@192.168.20.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.19:5060;branch=z9hG4bK31feac7e83
From: "5922261697" <sip:19@192.168.20.4>;tag=4b6f379d
To: <sip:user_ip@192.168.20.4>;tag=as607087e1
Call-ID: 5ac46a11150fb0a539162b705e7de357@192.168.20.19
CSeq: 802 ACK
Proxy-Authorization: Digest username="19",realm="asterisk",nonce="074a38f0",response="c4a564f66b8301e7014db12bd10539d3",uri="sip:user_ip@192.168.20.4",algorithm=MD5
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '5ac46a11150fb0a539162b705e7de357@192.168.20.19'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.20.19:5060:
OPTIONS sip:192.168.20.19 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.4:5060;branch=z9hG4bK7efedb96;rport
From: "Unknown" <sip:Unknown@192.168.20.4>;tag=as2338a578
To: <sip:192.168.20.19>
Contact: <sip:Unknown@192.168.20.4>
Call-ID: 663f2a2827b842461404c97d7250d005@192.168.20.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 17 Oct 2007 17:56:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk1*CLI> 
<-- SIP read from 192.168.20.19:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.4:5060;rport=5060;received=192.168.20.4;branch=z9hG4bK7efedb96
From: "Unknown" <sip:Unknown@192.168.20.4>;tag=as2338a578
To: <sip:192.168.20.19>;tag=04de48aa
Call-ID: 663f2a2827b842461404c97d7250d005@192.168.20.4
Contact: <sip:19@192.168.20.19:5060>
CSeq: 102 OPTIONS
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS
Content-Length: 0


--- (9 headers 0 lines) ---
Destroying call '663f2a2827b842461404c97d7250d005@192.168.20.4'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.20.19:5060:
OPTIONS sip:19@192.168.20.19:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.4:5060;branch=z9hG4bK19ced06c;rport
From: "Unknown" <sip:Unknown@192.168.20.4>;tag=as47308fda
To: <sip:19@192.168.20.19:5060>
Contact: <sip:Unknown@192.168.20.4>
Call-ID: 7053df037b453dcd4773479c748081b2@192.168.20.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 17 Oct 2007 17:56:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk1*CLI> 
<-- SIP read from 192.168.20.19:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.4:5060;rport=5060;received=192.168.20.4;branch=z9hG4bK19ced06c
From: "Unknown" <sip:Unknown@192.168.20.4>;tag=as47308fda
To: <sip:19@192.168.20.19:5060>;tag=1d749365
Call-ID: 7053df037b453dcd4773479c748081b2@192.168.20.4
Contact: <sip:19@192.168.20.19:5060>
CSeq: 102 OPTIONS
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS
Content-Length: 0