rendered paste body<-- SIP read from 85.95.72.181:5060:
INVITE sip:4912379@24.14.143.74 SIP/2.0
CSeq: 1 INVITE
Via: SIP/2.0/UDP 85.95.72.181:5060
From: <sip:0017737324583@85.95.72.181:5060>;tag=1204540719451887921109
Call-ID: 1887921109@85.95.72.181
To: <sip:4912379@24.14.143.74>
Contact: <sip:85.95.72.181:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 213
v=0
o=VoipSwitch 9522 9522 IN IP4 85.95.72.181
s=VoipSIP
i=Audio Session
c=IN IP4 85.95.72.181
t=0 0
m=audio 8522 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (9 headers 10 lines) ---
Using INVITE request as basis request - 1887921109@85.95.72.181
Sending to 85.95.72.181 : 5060 (NAT)
Found peer 'Voicelink'
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 85.95.72.181:8522
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Transmitting (NAT) to 85.95.72.181:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 85.95.72.181:5060;received=85.95.72.181
From: <sip:0017737324583@85.95.72.181:5060>;tag=1204540719451887921109
To: <sip:4912379@24.14.143.74>;tag=as715b02ef
Call-ID: 1887921109@85.95.72.181
CSeq: 1 INVITE
User-Agent: KillerPhone
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '1887921109@85.95.72.181'
rt-home*CLI>
<-- SIP read from 85.95.72.181:5060:
ACK sip:4912379@24.14.143.74 SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP 85.95.72.181:5060
From: <sip:0017737324583@85.95.72.181:5060>;tag=1204540719451887921109
Call-ID: 1887921109@85.95.72.181
To: <sip:4912379@24.14.143.74>;tag=as715b02ef
Contact: <sip:85.95.72.181:5060;transport=udp>
Content-Length: 0