rendered paste bodyDestroying call '77a6e1095722e5f86ef4db404016c8e3@71.41.69.70'
asterisk1*CLI>
<-- SIP read from 69.90.155.70:5060:
INVITE sip:s@71.41.69.70 SIP/2.0
Record-Route: <sip:765194@69.90.155.70;ftag=9f16bda61fe48348o0;lr=on>
Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bKc32e.6e474862.1
Via: SIP/2.0/UDP 192.168.1.114:5060;received=65.35.50.80;branch=z9hG4bK-daeb9f5c
From: Derrick Shand <sip:765427@fwd.pulver.com>;tag=9f16bda61fe48348o0
To: <sip:765194@fwd.pulver.com>
Call-ID: 9f9dbb7e-5709df50@192.168.1.114
CSeq: 102 INVITE
Max-Forwards: 69
Proxy-Authorization: Digest username="765427",realm="fwd.pulver.com",nonce="447e724729bb6037317ad33f53a4efb4d11a97fc",uri="sip:765194@fwd.pulver.com",algorithm=MD5,response="4c0398de6a8488f67eebd3d2c96cef34"
Contact: Derrick Shand <sip:765427@192.168.1.114:5060>
Expires: 240
User-Agent: Sipura/SPA2000-2.0.11(g)
Content-Length: 424
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 680503 680503 IN IP4 192.168.1.114
s=-
c=IN IP4 192.168.1.114
t=0 0
m=audio 16452 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (17 headers 19 lines)---
Using INVITE request as basis request - 9f9dbb7e-5709df50@192.168.1.114
Sending to 69.90.155.70 : 5060 (NAT)
Found peer 'test'
Reliably Transmitting (NAT) to 69.90.155.70:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bKc32e.6e474862.1;received=69.90.155.70
Via: SIP/2.0/UDP 192.168.1.114:5060;received=65.35.50.80;branch=z9hG4bK-daeb9f5c
From: Derrick Shand <sip:765427@fwd.pulver.com>;tag=9f16bda61fe48348o0
To: <sip:765194@fwd.pulver.com>;tag=as155b0653
Call-ID: 9f9dbb7e-5709df50@192.168.1.114
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@71.41.69.70>
Proxy-Authenticate: Digest realm="asterisk", nonce="450c3eeb"
Content-Length: 0
---
Scheduling destruction of call '9f9dbb7e-5709df50@192.168.1.114' in 15000 ms
asterisk1*CLI>
<-- SIP read from 69.90.155.70:5060:
ACK sip:s@71.41.69.70 SIP/2.0
Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bKc32e.6e474862.1
From: Derrick Shand <sip:765427@fwd.pulver.com>;tag=9f16bda61fe48348o0
Call-ID: 9f9dbb7e-5709df50@192.168.1.114
To: <sip:765194@fwd.pulver.com>;tag=as155b0653
CSeq: 102 ACK
User-Agent: Sip EXpress router(0.8.14-6 (i386/linux))
Content-Length: 0
--- (8 headers 0 lines)---
12 headers, 0 lines
Reliably Transmitting (NAT) to 69.90.155.70:5060:
OPTIONS sip:fwd.pulver.com SIP/2.0
Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK026971fc;rport
From: "asterisk" <sip:asterisk@71.41.69.70>;tag=as7b224b2d
To: <sip:fwd.pulver.com>
Contact: <sip:asterisk@71.41.69.70>
Call-ID: 6ee45b3a3467c7101865cd183d064397@71.41.69.70
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Jun 2006 04:39:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (NAT) to 69.90.155.70:5060:
OPTIONS sip:fwd.pulver.com SIP/2.0
Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK609c2c1d;rport
From: "asterisk" <sip:asterisk@71.41.69.70>;tag=as42e66054
To: <sip:fwd.pulver.com>
Contact: <sip:asterisk@71.41.69.70>
Call-ID: 5e6cb6ca3c3579e93d7540897456cb11@71.41.69.70
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Jun 2006 04:39:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (NAT) to 69.90.155.70:5060:
OPTIONS sip:fwd.pulver.com SIP/2.0
Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK5117f613;rport
From: "asterisk" <sip:asterisk@71.41.69.70>;tag=as70d38f70
To: <sip:fwd.pulver.com>
Contact: <sip:asterisk@71.41.69.70>
Call-ID: 4b84a2a73dcc6edc447e79ff2aefffbb@71.41.69.70
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Jun 2006 04:39:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 69.90.155.70:5060:
SIP/2.0 404 No such URI here
Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK026971fc;rport=5060
From: "asterisk" <sip:asterisk@71.41.69.70>;tag=as7b224b2d
To: <sip:fwd.pulver.com>;tag=cb2000b247d89723001a836145f3b053.2b05
Call-ID: 6ee45b3a3467c7101865cd183d064397@71.41.69.70
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '6ee45b3a3467c7101865cd183d064397@71.41.69.70'
asterisk1*CLI>
<-- SIP read from 69.90.155.70:5060:
SIP/2.0 404 No such URI here
Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK609c2c1d;rport=5060
From: "asterisk" <sip:asterisk@71.41.69.70>;tag=as42e66054
To: <sip:fwd.pulver.com>;tag=cb2000b247d89723001a836145f3b053.7bcc
Call-ID: 5e6cb6ca3c3579e93d7540897456cb11@71.41.69.70
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '5e6cb6ca3c3579e93d7540897456cb11@71.41.69.70'
asterisk1*CLI>
<-- SIP read from 69.90.155.70:5060:
SIP/2.0 404 No such URI here
Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK5117f613;rport=5060
From: "asterisk" <sip:asterisk@71.41.69.70>;tag=as70d38f70
To: <sip:fwd.pulver.com>;tag=cb2000b247d89723001a836145f3b053.60b9
Call-ID: 4b84a2a73dcc6edc447e79ff2aefffbb@71.41.69.70
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '4b84a2a73dcc6edc447e79ff2aefffbb@71.41.69.70'
12 headers, 0 lines
Reliably Transmitting (NAT) to 65.183.1.18:1262:
OPTIONS sip:202@10.10.10.7:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK7cde2765;rport
From: "asterisk" <sip:asterisk@71.41.69.70>;tag=as2b9e5e49
To: <sip:202@10.10.10.7:5060;user=phone;transport=udp>
Contact: <sip:asterisk@71.41.69.70>
Call-ID: 7c2fd6aa7294e27139d3f1a6138e24b0@71.41.69.70
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Jun 2006 04:39:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 65.183.1.18:1262:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 71.41.69.70:5060;branch=z9hG4bK7cde2765;rport
From: "asterisk" <sip:asterisk@71.41.69.70>;tag=as2b9e5e49
To: <sip:202@10.10.10.7:5060;user=phone;transport=udp>;tag=290228326
Call-ID: 7c2fd6aa7294e27139d3f1a6138e24b0@71.41.69.70
CSeq: 102 OPTIONS
Server: Cisco ATA 186 v3.1.0 atasip (040211A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 240
Content-Type: application/sdp
v=0
o=202 674187 674187 IN IP4 10.10.10.7
s=ATA186 Call
c=IN IP4 10.10.10.7
t=0 0
m=audio 0 RTP/AVP 0 8 4 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (10 headers 11 lines)---
Destroying call '7c2fd6aa7294e27139d3f1a6138e24b0@71.41.69.70'