Asterisk 1.4.2, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.2 currently running on officebox (pid = 2587)
Verbosity is at least 3
Really destroying SIP dialog 'NzQ3ZDBiMWRjZTQ1MWIzNDAwNDNkMjcwY2FiNGI2NWU.' Method: ACK
Really destroying SIP dialog 'NmQ4YjlmZmZjYTg4NzMwZTE3ODBlNGQzNzBjMGUyYzc.' Method: ACK
officebox*CLI>
<--- SIP read from 64.34.69.9:6968 --->
PUBLISH sip:100@64.34.139.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:6968;branch=z9hG4bK-d87543-075f5b7ddf293c5c-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:100@64.34.69.9:6968>
To: "Mark"<sip:100@64.34.139.98>
From: "Mark"<sip:100@64.34.139.98>;tag=5c0b2615
Call-ID: MWM0MWE4MjkxMGRiMjljM2ZkNmM2ZGFhN2RhNTBkNjQ.
CSeq: 1 PUBLISH
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/pidf+xml
User-Agent: X-Lite release 1006e stamp 34025
Event: presence
Content-Length: 456

<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='pres:100@64.34.139.98'><tuple id='teb184d0a'><status><basic>open</basic></status></tuple><dm:person id='pec096452'><rpid:activities><rpid:on-the-phone/></rpid:activities><dm:note>On the Phone</dm:note></dm:person></presence>
<------------->
--- (14 headers 1 lines) ---

<--- Transmitting (NAT) to 64.34.69.9:6968 --->
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 192.168.1.107:6968;branch=z9hG4bK-d87543-075f5b7ddf293c5c-1--d87543-;received=64.34.69.9;rport=6968
From: "Mark"<sip:100@64.34.139.98>;tag=5c0b2615
To: "Mark"<sip:100@64.34.139.98>;tag=as1a210e4a
Call-ID: MWM0MWE4MjkxMGRiMjljM2ZkNmM2ZGFhN2RhNTBkNjQ.
CSeq: 1 PUBLISH
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
officebox*CLI>
<--- SIP read from 64.34.69.9:6968 --->
INVITE sip:4162207455@64.34.139.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:6968;branch=z9hG4bK-d87543-1233e13ab3295559-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:100@64.34.69.9:6968>
To: "4162207455"<sip:4162207455@64.34.139.98>
From: "Mark"<sip:100@64.34.139.98>;tag=6b16413a
Call-ID: Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 285

v=0
o=- 8 2 IN IP4 192.168.1.107
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.107
t=0 0
m=audio 6792 RTP/AVP 0 8 101
a=alt:1 2 : UaG0F6Yu qgYXkui2 5.144.98.144 6792
a=alt:2 1 : 0oPLoIp9 BCSFSEsa 192.168.1.107 6792
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 11 lines) ---
Sending to 64.34.69.9 : 6968 (NAT)
Using INVITE request as basis request - Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.
Found peer '100'

<--- Reliably Transmitting (NAT) to 64.34.69.9:6968 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.107:6968;branch=z9hG4bK-d87543-1233e13ab3295559-1--d87543-;received=64.34.69.9;rport=6968
From: "Mark"<sip:100@64.34.139.98>;tag=6b16413a
To: "4162207455"<sip:4162207455@64.34.139.98>;tag=as779bc219
Call-ID: Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.
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="3f8835e1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.' in 32000 ms (Method: INVITE)
officebox*CLI>
<--- SIP read from 64.34.69.9:6968 --->
ACK sip:4162207455@64.34.139.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:6968;branch=z9hG4bK-d87543-1233e13ab3295559-1--d87543-;rport
To: "4162207455"<sip:4162207455@64.34.139.98>;tag=as779bc219
From: "Mark"<sip:100@64.34.139.98>;tag=6b16413a
Call-ID: Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
officebox*CLI>
<--- SIP read from 64.34.69.9:6968 --->
INVITE sip:4162207455@64.34.139.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:6968;branch=z9hG4bK-d87543-3642b546ed1b6526-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:100@64.34.69.9:6968>
To: "4162207455"<sip:4162207455@64.34.139.98>
From: "Mark"<sip:100@64.34.139.98>;tag=6b16413a
Call-ID: Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="100",realm="asterisk",nonce="3f8835e1",uri="sip:4162207455@64.34.139.98",response="fe9410479023d7138d4be639ffb5d38f",algorithm=MD5
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 285

v=0
o=- 8 2 IN IP4 192.168.1.107
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.107
t=0 0
m=audio 6792 RTP/AVP 0 8 101
a=alt:1 2 : UaG0F6Yu qgYXkui2 5.144.98.144 6792
a=alt:2 1 : 0oPLoIp9 BCSFSEsa 192.168.1.107 6792
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 11 lines) ---
Sending to 64.34.69.9 : 6968 (NAT)
Using INVITE request as basis request - Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.
Found peer '100'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.107:6792
Found description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0xc (ulaw|alaw)/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 192.168.1.107:6792
Looking for 4162207455 in numberplan-custom-1 (domain 64.34.139.98)

<--- Reliably Transmitting (NAT) to 64.34.69.9:6968 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.107:6968;branch=z9hG4bK-d87543-3642b546ed1b6526-1--d87543-;received=64.34.69.9;rport=6968
From: "Mark"<sip:100@64.34.139.98>;tag=6b16413a
To: "4162207455"<sip:4162207455@64.34.139.98>;tag=as779bc219
Call-ID: Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.' in 32000 ms (Method: INVITE)
officebox*CLI>
<--- SIP read from 64.34.69.9:6968 --->
ACK sip:4162207455@64.34.139.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:6968;branch=z9hG4bK-d87543-3642b546ed1b6526-1--d87543-;rport
To: "4162207455"<sip:4162207455@64.34.139.98>;tag=as779bc219
From: "Mark"<sip:100@64.34.139.98>;tag=6b16413a
Call-ID: Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
officebox*CLI>
<--- SIP read from 64.34.69.9:6968 --->
PUBLISH sip:100@64.34.139.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:6968;branch=z9hG4bK-d87543-165ddb2c7a6b7c5a-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:100@64.34.69.9:6968>
To: "Mark"<sip:100@64.34.139.98>
From: "Mark"<sip:100@64.34.139.98>;tag=d3157170
Call-ID: OGZhZjcxMjhlOGZjMDczMjEwZDkwYzM1NzI2ODg0MzI.
CSeq: 1 PUBLISH
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/pidf+xml
User-Agent: X-Lite release 1006e stamp 34025
Event: presence
Content-Length: 420

<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='pres:100@64.34.139.98'><tuple id='teb184d0a'><status><basic>open</basic></status></tuple><dm:person id='pec096452'><rpid:activities><rpid:unknown/></rpid:activities></dm:person></presence>
<------------->
--- (14 headers 1 lines) ---

<--- Transmitting (NAT) to 64.34.69.9:6968 --->
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 192.168.1.107:6968;branch=z9hG4bK-d87543-165ddb2c7a6b7c5a-1--d87543-;received=64.34.69.9;rport=6968
From: "Mark"<sip:100@64.34.139.98>;tag=d3157170
To: "Mark"<sip:100@64.34.139.98>;tag=as6b9888ab
Call-ID: OGZhZjcxMjhlOGZjMDczMjEwZDkwYzM1NzI2ODg0MzI.
CSeq: 1 PUBLISH
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
officebox*CLI>
<--- SIP read from 64.34.69.9:6968 --->



<------------->
--- (0 headers 1 lines) ---
Really destroying SIP dialog 'Njg3NzM3Y2NlZmUyMWM2NGFmNDFhMzJkNDNlNGJjYWQ.' Method: ACK
officebox*CLI>
<--- SIP read from 64.34.69.9:6968 --->



<------------->
--- (0 headers 1 lines) ---
officebox*CLI>
<--- SIP read from 64.34.69.9:6968 --->



<------------->
--- (0 headers 1 lines) ---