rendered paste body<-- SIP read from 64.182.154.100:5060:
INVITE sip:8778665208@198.211.206.173 SIP/2.0
Max-Forwards: 19
Via: SIP/2.0/UDP 64.182.154.100:5060;rport;branch=z9hG4bK1347147116
From: "8175003884" <sip:8175003884@64.182.154.100>;tag=363033930
To: <sip:8778665208@198.211.206.173>
Call-ID: 877851817@64.182.154.100
CSeq: 15930 INVITE
User-Agent: Sur-Tel Communications CTG 6.5 ORG
Contact: <sip:8175003884@64.182.154.100:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 234
v=0
o=yate 1170554979 1170554979 IN IP4 64.182.154.100
s=SIP Call
c=IN IP4 64.182.154.100
t=0 0
m=audio 15888 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
--- (12 headers 10 lines)---
Using INVITE request as basis request - 877851817@64.182.154.100
Sending to 64.182.154.100 : 5060 (NAT)
Found peer 'netiqsysin'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 64.182.154.100:15888
Found description format PCMU
Found description format PCMA
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 8778665208 in default (domain 198.211.206.173)
Reliably Transmitting (no NAT) to 64.182.154.100:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 64.182.154.100:5060;rport;branch=z9hG4bK1347147116;received=64.182.154.100
From: "8175003884" <sip:8175003884@64.182.154.100>;tag=363033930
To: <sip:8778665208@198.211.206.173>;tag=as6144032a
Call-ID: 877851817@64.182.154.100
CSeq: 15930 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8778665208@198.211.206.173>
Content-Length: 0
---
zap*CLI>
<-- SIP read from 64.182.154.100:5060:
ACK sip:8778665208@198.211.206.173 SIP/2.0
Via: SIP/2.0/UDP 64.182.154.100:5060;rport;branch=z9hG4bK1347147116
From: "8175003884" <sip:8175003884@64.182.154.100>;tag=363033930
To: <sip:8778665208@198.211.206.173>;tag=as6144032a
Call-ID: 877851817@64.182.154.100
CSeq: 15930 ACK
Max-Forwards: 19
Contact: <sip:8175003884@64.182.154.100:5060>
User-Agent: Sur-Tel Communications CTG 6.5 ORG
Content-Length: 0
--- (10 headers 0 lines)---
Destroying call '877851817@64.182.154.100'
zap*CLI>
<-- SIP read from 64.182.154.100:5060:
INVITE sip:8778665208@198.211.206.173 SIP/2.0
Max-Forwards: 19
Via: SIP/2.0/UDP 64.182.154.100:5060;rport;branch=z9hG4bK1121922176
From: "8175003884" <sip:8175003884@64.182.154.100>;tag=528990268
To: <sip:8778665208@198.211.206.173>
Call-ID: 1381562573@64.182.154.100
CSeq: 15931 INVITE
User-Agent: Sur-Tel Communications CTG 6.5 ORG
Contact: <sip:8175003884@64.182.154.100:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 234
v=0
o=yate 1170554979 1170554979 IN IP4 64.182.154.100
s=SIP Call
c=IN IP4 64.182.154.100
t=0 0
m=audio 15076 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
--- (12 headers 10 lines)---
Using INVITE request as basis request - 1381562573@64.182.154.100
Sending to 64.182.154.100 : 5060 (NAT)
Found peer 'netiqsysin'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 64.182.154.100:15076
Found description format PCMU
Found description format PCMA
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 8778665208 in default (domain 198.211.206.173)
Reliably Transmitting (no NAT) to 64.182.154.100:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 64.182.154.100:5060;rport;branch=z9hG4bK1121922176;received=64.182.154.100
From: "8175003884" <sip:8175003884@64.182.154.100>;tag=528990268
To: <sip:8778665208@198.211.206.173>;tag=as62eac13a
Call-ID: 1381562573@64.182.154.100
CSeq: 15931 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8778665208@198.211.206.173>
Content-Length: 0
---
zap*CLI>
<-- SIP read from 64.182.154.100:5060:
ACK sip:8778665208@198.211.206.173 SIP/2.0
Via: SIP/2.0/UDP 64.182.154.100:5060;rport;branch=z9hG4bK1121922176
From: "8175003884" <sip:8175003884@64.182.154.100>;tag=528990268
To: <sip:8778665208@198.211.206.173>;tag=as62eac13a
Call-ID: 1381562573@64.182.154.100
CSeq: 15931 ACK
Max-Forwards: 19
Contact: <sip:8175003884@64.182.154.100:5060>
User-Agent: Sur-Tel Communications CTG 6.5 ORG
Content-Length: 0
--- (10 headers 0 lines)---
Destroying call '1381562573@64.182.154.100'