All pastes #460194 Raw Edit

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#460194 ·published 2007-04-26 22:17 UTC
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Proxy-Authorization: Digest username="8868231", realm="asterisk", algorithm=MD5, uri="sip:13134453623@72.4.160.99", nonce="0088e93c", response="132cd5d45f1eec878b713ce46c9fb727"
Call-ID: 0a3d3337f388d3d5@192.168.1.100
CSeq: 11506 INVITE
User-Agent: Grandstream GXP2000 1.1.0.5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 257

v=0
o=8868231 8000 8001 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 209.19.88.240
t=0 0
m=audio 5004 RTP/AVP 4 8 18 3 0
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=ptime:20

--- (14 headers 13 lines) ---
Using INVITE request as basis request - 0a3d3337f388d3d5@192.168.1.100
Sending to 209.19.88.240 : 5062 (NAT)
Found user '8868231'
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 0
Peer audio RTP is at port 209.19.88.240:5004
Found description format G723
Found description format PCMA
Found description format G729
Found description format GSM
Found description format PCMU
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 13134453623 in a2billing (domain 72.4.160.99)
Reliably Transmitting (NAT) to 209.19.88.240:5062:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bK48cf6ff105f00a6d;received=209.19.88.240
From: "3138875702" <sip:8868231@72.4.160.99>;tag=ed140170ea65a119
To: <sip:13134453623@72.4.160.99>;tag=as76d864d4
Call-ID: 0a3d3337f388d3d5@192.168.1.100
CSeq: 11506 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
localhost*CLI>
<-- SIP read from 209.19.88.240:5062:
ACK sip:13134453623@72.4.160.99 SIP/2.0
Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bK48cf6ff105f00a6d
From: "3138875702" <sip:8868231@72.4.160.99>;tag=ed140170ea65a119
To: <sip:13134453623@72.4.160.99>;tag=as76d864d4
Contact: <sip:8868231@209.19.88.240:5062>
Proxy-Authorization: Digest username="8868231", realm="asterisk", algorithm=MD5, uri="sip:13134453623@72.4.160.99", nonce="0088e93c", response="0ca166d65e8fc8abdcda17ffe76a7d17"
Call-ID: 0a3d3337f388d3d5@192.168.1.100
CSeq: 11506 ACK
User-Agent: Grandstream GXP2000 1.1.0.5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0


--- (12 headers 0 lines) ---
Destroying call '0a3d3337f388d3d5@192.168.1.100'
localhost*CLI> sip no debug
SIP Debugging Disabled
localhost*CLI> \
localhost*CLI> !
[root@localhost ~]# clear
[root@localhost ~]# clear
[root@localhost ~]# asterisk -vr
Asterisk 1.2.18, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.18 currently running on localhost (pid = 1971)
Verbosity is at least 1
localhost*CLI> sip debug
SIP Debugging enabled
localhost*CLI>
<-- SIP read from 209.19.88.240:5062:
INVITE sip:13134453623@72.4.160.99 SIP/2.0
Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bKdeec3dbf1c7d07f4
From: "3138875702" <sip:8868231@72.4.160.99>;tag=62251498eda64ee6
To: <sip:13134453623@72.4.160.99>
Contact: <sip:8868231@209.19.88.240:5062>
Supported: replaces, timer
Call-ID: dc4721988f5a0589@192.168.1.100
CSeq: 24790 INVITE
User-Agent: Grandstream GXP2000 1.1.0.5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 257

v=0
o=8868231 8000 8000 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 209.19.88.240
t=0 0
m=audio 5004 RTP/AVP 4 8 18 3 0
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=ptime:20

--- (13 headers 13 lines) ---
Using INVITE request as basis request - dc4721988f5a0589@192.168.1.100
Sending to 209.19.88.240 : 5062 (non-NAT)
Reliably Transmitting (NAT) to 209.19.88.240:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bKdeec3dbf1c7d07f4;received=209.19.88.240
From: "3138875702" <sip:8868231@72.4.160.99>;tag=62251498eda64ee6
To: <sip:13134453623@72.4.160.99>;tag=as64dbe8d0
Call-ID: dc4721988f5a0589@192.168.1.100
CSeq: 24790 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17ce0a18"
Content-Length: 0


---
Scheduling destruction of call 'dc4721988f5a0589@192.168.1.100' in 15000 ms
Found user '8868231'
localhost*CLI>
<-- SIP read from 209.19.88.240:5062:
ACK sip:13134453623@72.4.160.99 SIP/2.0
Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bKdeec3dbf1c7d07f4
From: "3138875702" <sip:8868231@72.4.160.99>;tag=62251498eda64ee6
To: <sip:13134453623@72.4.160.99>;tag=as64dbe8d0
Contact: <sip:8868231@209.19.88.240:5062>
Call-ID: dc4721988f5a0589@192.168.1.100
CSeq: 24790 ACK
User-Agent: Grandstream GXP2000 1.1.0.5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0


--- (11 headers 0 lines) ---
localhost*CLI>
<-- SIP read from 209.19.88.240:5062:
INVITE sip:13134453623@72.4.160.99 SIP/2.0
Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bK919f159680a0db66
From: "3138875702" <sip:8868231@72.4.160.99>;tag=62251498eda64ee6
To: <sip:13134453623@72.4.160.99>
Contact: <sip:8868231@209.19.88.240:5062>
Supported: replaces, timer
Proxy-Authorization: Digest username="8868231", realm="asterisk", algorithm=MD5, uri="sip:13134453623@72.4.160.99", nonce="17ce0a18", response="3a590a215509ad6cc64a7592852cf29a"
Call-ID: dc4721988f5a0589@192.168.1.100
CSeq: 24791 INVITE
User-Agent: Grandstream GXP2000 1.1.0.5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 257

v=0
o=8868231 8000 8001 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 209.19.88.240
t=0 0
m=audio 5004 RTP/AVP 4 8 18 3 0
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=ptime:20

--- (14 headers 13 lines) ---
Using INVITE request as basis request - dc4721988f5a0589@192.168.1.100
Sending to 209.19.88.240 : 5062 (NAT)
Found user '8868231'
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 0
Peer audio RTP is at port 209.19.88.240:5004
Found description format G723
Found description format PCMA
Found description format G729
Found description format GSM
Found description format PCMU
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 13134453623 in a2billing (domain 72.4.160.99)
Reliably Transmitting (NAT) to 209.19.88.240:5062:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bK919f159680a0db66;received=209.19.88.240
From: "3138875702" <sip:8868231@72.4.160.99>;tag=62251498eda64ee6
To: <sip:13134453623@72.4.160.99>;tag=as64dbe8d0
Call-ID: dc4721988f5a0589@192.168.1.100
CSeq: 24791 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
localhost*CLI>
<-- SIP read from 209.19.88.240:5062:
ACK sip:13134453623@72.4.160.99 SIP/2.0
Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bK919f159680a0db66
From: "3138875702" <sip:8868231@72.4.160.99>;tag=62251498eda64ee6
To: <sip:13134453623@72.4.160.99>;tag=as64dbe8d0
Contact: <sip:8868231@209.19.88.240:5062>
Proxy-Authorization: Digest username="8868231", realm="asterisk", algorithm=MD5, uri="sip:13134453623@72.4.160.99", nonce="17ce0a18", response="90198d838bfb6b5df37ce3095815f5e3"
Call-ID: dc4721988f5a0589@192.168.1.100
CSeq: 24791 ACK
User-Agent: Grandstream GXP2000 1.1.0.5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0


--- (12 headers 0 lines) ---
Destroying call 'dc4721988f5a0589@192.168.1.100'
localhost*CLI> sip no debug
SIP Debugging Disabled