rendered paste bodyREGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.24.35.208:5060:
REGISTER sip:chi2-sr1-ha.starnetusa.net SIP/2.0
Via: SIP/2.0/UDP 71.162.124.10:5060;branch=z9hG4bK13cfedd8;rport
From: <sip:58888@chi2-sr1-ha.starnetusa.net>;tag=as47aa3900
To: <sip:58888@chi2-sr1-ha.starnetusa.net>
Call-ID: 312cbf767f2d5a380329c3282d328041@127.0.0.1
CSeq: 404 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="58888", realm="chi2-sr1-ha.starnetusa.net", algorithm=MD5, uri="sip:chi2-sr1-ha.starnetusa.net", nonce="4500da950a2a23719eaa588b54479853c9a1fa0d", response="5051c46e67c2fa2971a4d09c01b1bac4", opaque=""
Expires: 120
Contact: <sip:7814180525@71.162.124.10>
Event: registration
Content-Length: 0
---
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.24.35.208:5060:
REGISTER sip:chi2-sr1-ha.starnetusa.net SIP/2.0
Via: SIP/2.0/UDP 71.162.124.10:5060;branch=z9hG4bK604da42c;rport
From: <sip:58887@chi2-sr1-ha.starnetusa.net>;tag=as4876702c
To: <sip:58887@chi2-sr1-ha.starnetusa.net>
Call-ID: 4f90f3f51314e4526a9ef1d239de5119@127.0.0.1
CSeq: 404 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="58887", realm="chi2-sr1-ha.starnetusa.net", algorithm=MD5, uri="sip:chi2-sr1-ha.starnetusa.net", nonce="4500da950a2a23719eaa588b54479853c9a1fa0d", response="650e07aaec760385fcf383f73796a54c", opaque=""
Expires: 120
Contact: <sip:7814180850@71.162.124.10>
Event: registration
Content-Length: 0
---
<-- SIP read from 64.24.35.208:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 71.162.124.10:5060;branch=z9hG4bK13cfedd8;rport=5060
From: <sip:58888@chi2-sr1-ha.starnetusa.net>;tag=as47aa3900
To: <sip:58888@chi2-sr1-ha.starnetusa.net>;tag=70d7056f49d78fbf344909f489af4c1c.da5e
Call-ID: 312cbf767f2d5a380329c3282d328041@127.0.0.1
CSeq: 404 REGISTER
Contact: <sip:7814180525@71.162.124.10>;expires=60
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 64.24.35.208:5060 "Noisy feedback tells: pid=12778 req_src_ip=71.162.124.10 req_src_port=5060 in_uri=sip:chi2-sr1-ha.starnetusa.net out_uri=sip:chi2-sr1-ha.starnetusa.net via_cnt==1"
--- (10 headers 0 lines)---
Scheduling destruction of call '312cbf767f2d5a380329c3282d328041@127.0.0.1' in 32000 ms
asterisk1*CLI>
<-- SIP read from 64.24.35.208:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 71.162.124.10:5060;branch=z9hG4bK604da42c;rport=5060
From: <sip:58887@chi2-sr1-ha.starnetusa.net>;tag=as4876702c
To: <sip:58887@chi2-sr1-ha.starnetusa.net>;tag=70d7056f49d78fbf344909f489af4c1c.b810
Call-ID: 4f90f3f51314e4526a9ef1d239de5119@127.0.0.1
CSeq: 404 REGISTER
Contact: <sip:7814180850@71.162.124.10>;expires=60
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 64.24.35.208:5060 "Noisy feedback tells: pid=12776 req_src_ip=71.162.124.10 req_src_port=5060 in_uri=sip:chi2-sr1-ha.starnetusa.net out_uri=sip:chi2-sr1-ha.starnetusa.net via_cnt==1"
--- (10 headers 0 lines)---
Scheduling destruction of call '4f90f3f51314e4526a9ef1d239de5119@127.0.0.1' in 32000 ms
asterisk1*CLI>
<-- SIP read from 64.24.35.208:5060:
INVITE sip:7814180525@71.162.124.10 SIP/2.0
Via: SIP/2.0/UDP 64.24.35.208;branch=0
Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK78131873;rport=5060
From: "7819323036" <sip:7819323036@chi2-sr1-ha>;tag=as036e032f
To: <sip:58888@64.24.35.208>
Contact: <sip:7819323036@64.24.35.73>
Call-ID: 51549d762b58608b44493fe72af31f6b@chi2-sr1-ha
CSeq: 102 INVITE
User-Agent: US LEC VOIP
Max-Forwards: 16
Date: Fri, 08 Sep 2006 02:49:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 10925 10925 IN IP4 64.24.35.73
s=session
c=IN IP4 64.24.35.73
t=0 0
m=audio 26800 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (14 headers 12 lines)---
Using INVITE request as basis request - 51549d762b58608b44493fe72af31f6b@chi2-sr1-ha
Sending to 64.24.35.208 : 5060 (non-NAT)
Found peer 'Secondary Line'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 64.24.35.73:26800
Found description format G729
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 7814180525 in from-sip-external (domain 71.162.124.10)
list_route: hop: <sip:7819323036@64.24.35.73>
Transmitting (NAT) to 64.24.35.208:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.24.35.208;branch=0;received=64.24.35.208
Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK78131873;rport=5060
From: "7819323036" <sip:7819323036@chi2-sr1-ha>;tag=as036e032f
To: <sip:58888@64.24.35.208>
Call-ID: 51549d762b58608b44493fe72af31f6b@chi2-sr1-ha
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7814180525@71.162.124.10>
Content-Length: 0
---
We're at 71.162.124.10 port 17146
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.24.35.208:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.24.35.208;branch=0;received=64.24.35.208
Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK78131873;rport=5060
From: "7819323036" <sip:7819323036@chi2-sr1-ha>;tag=as036e032f
To: <sip:58888@64.24.35.208>;tag=as6fd4f003
Call-ID: 51549d762b58608b44493fe72af31f6b@chi2-sr1-ha
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7814180525@71.162.124.10>
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 2726 2726 IN IP4 71.162.124.10
s=session
c=IN IP4 71.162.124.10
t=0 0
m=audio 17146 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #1 (NAT) to 64.24.35.208:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.24.35.208;branch=0;received=64.24.35.208
Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK78131873;rport=5060
From: "7819323036" <sip:7819323036@chi2-sr1-ha>;tag=as036e032f
To: <sip:58888@64.24.35.208>;tag=as6fd4f003
Call-ID: 51549d762b58608b44493fe72af31f6b@chi2-sr1-ha
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7814180525@71.162.124.10>
ontent-Type: application/sdp
Content-Length: 216
v=0
o=root 2726 2726 IN IP4 71.162.124.10
s=session
c=IN IP4 71.162.124.10
t=0 0
m=audio 17146 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #2 (NAT) to 64.24.35.208:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.24.35.208;branch=0;received=64.24.35.208
Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK78131873;rport=5060
From: "7819323036" <sip:7819323036@chi2-sr1-ha>;tag=as036e032f
To: <sip:58888@64.24.35.208>;tag=as6fd4f003
Call-ID: 51549d762b58608b44493fe72af31f6b@chi2-sr1-ha
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7814180525@71.162.124.10>
ontent-Type: application/sdp
Content-Length: 216
v=0
o=root 2726 2726 IN IP4 71.162.124.10
s=session
c=IN IP4 71.162.124.10
t=0 0
m=audio 17146 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #3 (NAT) to 64.24.35.208:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.24.35.208;branch=0;received=64.24.35.208
Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK78131873;rport=5060
From: "7819323036" <sip:7819323036@chi2-sr1-ha>;tag=as036e032f
To: <sip:58888@64.24.35.208>;tag=as6fd4f003
Call-ID: 51549d762b58608b44493fe72af31f6b@chi2-sr1-ha
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7814180525@71.162.124.10>
ontent-Type: application/sdp
Content-Length: 216
v=0
o=root 2726 2726 IN IP4 71.162.124.10
s=session
c=IN IP4 71.162.124.10
t=0 0
m=audio 17146 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #4 (NAT) to 64.24.35.208:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.24.35.208;branch=0;received=64.24.35.208
Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK78131873;rport=5060
From: "7819323036" <sip:7819323036@chi2-sr1-ha>;tag=as036e032f
To: <sip:58888@64.24.35.208>;tag=as6fd4f003
Call-ID: 51549d762b58608b44493fe72af31f6b@chi2-sr1-ha
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7814180525@71.162.124.10>
ontent-Type: application/sdp
Content-Length: 216
v=0
o=root 2726 2726 IN IP4 71.162.124.10
s=session
c=IN IP4 71.162.124.10
t=0 0
m=audio 17146 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #5 (NAT) to 64.24.35.208:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.24.35.208;branch=0;received=64.24.35.208
Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK78131873;rport=5060
From: "7819323036" <sip:7819323036@chi2-sr1-ha>;tag=as036e032f
To: <sip:58888@64.24.35.208>;tag=as6fd4f003
Call-ID: 51549d762b58608b44493fe72af31f6b@chi2-sr1-ha
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7814180525@71.162.124.10>
ontent-Type: application/sdp
Content-Length: 216
v=0
o=root 2726 2726 IN IP4 71.162.124.10
s=session
c=IN IP4 71.162.124.10
t=0 0
m=audio 17146 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #6 (NAT) to 64.24.35.208:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.24.35.208;branch=0;received=64.24.35.208
Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK78131873;rport=5060
From: "7819323036" <sip:7819323036@chi2-sr1-ha>;tag=as036e032f
To: <sip:58888@64.24.35.208>;tag=as6fd4f003
Call-ID: 51549d762b58608b44493fe72af31f6b@chi2-sr1-ha
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7814180525@71.162.124.10>
ontent-Type: application/sdp
Content-Length: 216
v=0
o=root 2726 2726 IN IP4 71.162.124.10
s=session
c=IN IP4 71.162.124.10
t=0 0
m=audio 17146 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Destroying call '312cbf767f2d5a380329c3282d328041@127.0.0.1'
Destroying call '4f90f3f51314e4526a9ef1d239de5119@127.0.0.1'
asterisk1*CLI>