rendered paste body<-- Sip read from 65.244.210.40:5060:UDP
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 65.244.210.42:5060;branch=z9hG4bK3eb6d56f
Content-Length: 0
Contact: <sip:6248@65.244.210.40:5060>
Call-ID: 0dca1c84587177dc568ed41626300257@65.244.210.42
CSeq: 102 INVITE
From: "6092286200"<sip:6092286200@65.244.210.42>;tag=as2b172f94
To: "unknown"<sip:6248@65.244.210.40:5060>;tag=21245431328826
--- (8 headers 0 lines)---
UDP write: fd 12
UDP write: fd 12
asterisk1*CLI>
<-- Sip read from 65.244.210.40:5060:UDP
SIP/2.0 200 OK
Via: SIP/2.0/UDP 65.244.210.42:5060;branch=z9hG4bK3eb6d56f
Content-Length: 221
Contact: <sip:6248@65.244.210.40:5060>
Call-ID: 0dca1c84587177dc568ed41626300257@65.244.210.42
Content-Type: application/sdp
CSeq: 102 INVITE
From: "6092286200"<sip:6092286200@65.244.210.42>;tag=as2b172f94
To: "unknown"<sip:6248@65.244.210.40:5060>;tag=21245431328826
v=0
o=- 3367835603 3367835603 IN IP4 65.244.210.40
s=SJphone
c=IN IP4 65.244.210.40
t=0 0
a=direction:active
m=audio 49166 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
--- (9 headers 10 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 65.244.210.40:49166
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:6248@65.244.210.40:5060>
set_destination: Parsing <sip:6248@65.244.210.40:5060> for address/port/transport to send to
Transmitting (no NAT) to 65.244.210.40:5060:UDP
with fd 12
ACK sip:6248@65.244.210.40:5060 SIP/2.0
Via: SIP/2.0/UDP 65.244.210.42:5060;branch=z9hG4bK662d668c
From: "6092286200" <sip:6092286200@65.244.210.42>;tag=as2b172f94
To: <sip:6248@65.244.210.40:5060>;tag=21245431328826
Contact: <sip:6092286200@65.244.210.42;transport=UDP>
Call-ID: 0dca1c84587177dc568ed41626300257@65.244.210.42
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
UDP write: fd 12
UDP write: fd 12
UDP write: fd 12
UDP write: fd 12
UDP write: fd 12
UDP write: fd 12
asterisk1*CLI>
<-- Sip read from 65.244.210.40:5060:UDP
OPTIONS sip:65.244.210.42 SIP/2.0
Via: SIP/2.0/UDP 65.244.210.40;rport;branch=z9hG4bK41f4d228000000104512995900005aa600000122
Content-Length: 0
Call-ID: 7BB675C6-7E34-425B-9987-3C85C6172328@65.244.210.40
CSeq: 80 OPTIONS
From: <sip:6248@65.244.210.42>;tag=21246053216900
Max-Forwards: 70
To: <sip:65.244.210.42>
--- (8 headers 0 lines)---
Looking for s in from-sip-external (domain 65.244.210.42)
Transmitting (no NAT) to 65.244.210.40:5060:UDP
with fd 12
SIP/2.0 200 OK
Via: SIP/2.0/UDP 65.244.210.40;rport;branch=z9hG4bK41f4d228000000104512995900005aa600000122;received=65.244.210.40
From: <sip:6248@65.244.210.42>;tag=21246053216900
To: <sip:65.244.210.42>;tag=as7fe94b6a
Call-ID: 7BB675C6-7E34-425B-9987-3C85C6172328@65.244.210.40
CSeq: 80 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:65.244.210.42;transport=UDP>
Accept: application/sdp
Content-Length: 0
---
UDP write: fd 12
Destroying call '7BB675C6-7E34-425B-9987-3C85C6172328@65.244.210.40'
UDP write: fd 12
UDP write: fd 12
asterisk1*CLI>
<-- Sip read from 65.244.210.40:5060:UDP
BYE sip:6092286200@65.244.210.42;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.244.210.40;rport;branch=z9hG4bK41f4d2280000005c4512995a00005fcb00000124
Content-Length: 0
Call-ID: 0dca1c84587177dc568ed41626300257@65.244.210.42
CSeq: 1 BYE
From: "unknown"<sip:6248@65.244.210.40:5060>;tag=21245431328826
Max-Forwards: 70
To: <sip:6092286200@65.244.210.42>;tag=as2b172f94
--- (8 headers 0 lines)---
Sending to 65.244.210.40: 5060; transport UDP (NAT)
Transmitting (NAT) to 65.244.210.40:5060:UDP
with fd 12
SIP/2.0 200 OK
Via: SIP/2.0/UDP 65.244.210.40;rport;branch=z9hG4bK41f4d2280000005c4512995a00005fcb00000124;received=65.244.210.40
From: "unknown"<sip:6248@65.244.210.40:5060>;tag=21245431328826
To: <sip:6092286200@65.244.210.42>;tag=as2b172f94
Call-ID: 0dca1c84587177dc568ed41626300257@65.244.210.42
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6092286200@65.244.210.42;transport=UDP>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
UDP write: fd 12
UDP write: fd 12
Destroying call '0dca1c84587177dc568ed41626300257@65.244.210.42'