*CLI>     -- Accepting UNAUTHENTICATED call from 209.217.98.230:
       > requested format = ulaw,
       > requested prefs = (ulaw|gsm),
       > actual format = ulaw,
       > host prefs = (ulaw|gsm),
       > priority = mine
    -- Executing [6136274820@6136274820-in:1] Dial("IAX2/209.217.98.230:4569-7549", "SIP/6136274820@IVR1") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 172.16.2.185 port 11528
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.2.190:5060:
INVITE sip:6136274820@172.16.2.190 SIP/2.0
Via: SIP/2.0/TCP 172.16.2.185:5060;branch=z9hG4bK500d5fb5;rport
Max-Forwards: 70
From: "Unknown Name" <sip:6137959112@172.16.2.185>;tag=as43494201
To: <sip:6136274820@172.16.2.190>
Contact: <sip:6137959112@172.16.2.185:5060;transport=TCP>
Call-ID: 3f2d24a02c67b01212d53b847642f08b@172.16.2.185
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta8
Date: Tue, 29 Apr 2008 16:30:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1580938582 1580938582 IN IP4 172.16.2.185
s=Asterisk PBX 1.6.0-beta8
c=IN IP4 172.16.2.185
t=0 0
m=audio 11528 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 6136274820@IVR1

<--- SIP read from TCP://172.16.2.190:5060 --->
SIP/2.0 100 Trying
FROM: "Unknown Name"<sip:6137959112@172.16.2.185>;tag=as43494201
TO: <sip:6136274820@172.16.2.190>
CSEQ: 102 INVITE
CALL-ID: 3f2d24a02c67b01212d53b847642f08b@172.16.2.185
VIA: SIP/2.0/TCP 172.16.2.185:5060;branch=z9hG4bK500d5fb5;rport
CONTENT-LENGTH: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from TCP://172.16.2.190:5060 --->
SIP/2.0 302 Moved Temporarily
FROM: "Unknown Name"<sip:6137959112@172.16.2.185>;tag=as43494201
TO: <sip:6136274820@172.16.2.190>;tag=e64e5385f4
CSEQ: 102 INVITE
CALL-ID: 3f2d24a02c67b01212d53b847642f08b@172.16.2.185
VIA: SIP/2.0/TCP 172.16.2.185:5060;branch=z9hG4bK500d5fb5;rport
CONTACT: <sip:6136274820@172.16.2.190:5070;transport=Tcp;maddr=172.16.2.190;x-mss-call-id=3f2d24a02c67b01212d53b847642f08b%40172.16.2.185>
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0


<------------->
--- (9 headers 0 lines) ---
    -- Got SIP response 302 "Moved Temporarily" back from 172.16.2.190
Transmitting (no NAT) to 172.16.2.190:5060:
ACK sip:6136274820@172.16.2.190 SIP/2.0
Via: SIP/2.0/TCP 172.16.2.185:5060;branch=z9hG4bK500d5fb5;rport
Max-Forwards: 70
From: "Unknown Name" <sip:6137959112@172.16.2.185>;tag=as43494201
To: <sip:6136274820@172.16.2.190>;tag=e64e5385f4
Contact: <sip:6137959112@172.16.2.185:5060;transport=TCP>
Call-ID: 3f2d24a02c67b01212d53b847642f08b@172.16.2.185
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0-beta8
Content-Length: 0


---
    -- Now forwarding IAX2/209.217.98.230:4569-7549 to 'SIP/6136274820@172.16.2.190:5070' (thanks to SIP/IVR1-081e28b8)
  == Using SIP RTP CoS mark 5
Audio is at 172.16.2.185 port 13422
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.2.190:5070:
INVITE sip:6136274820@172.16.2.190:5070 SIP/2.0
Via: SIP/2.0/UDP 172.16.2.185:5060;branch=z9hG4bK56ce0e9b;rport
Max-Forwards: 70
From: "Unknown Name" <sip:6137959112@172.16.2.185>;tag=as1edf1b2c
To: <sip:6136274820@172.16.2.190:5070>
Contact: <sip:6137959112@172.16.2.185>
Call-ID: 5dd1a7f12d5e6b201df7f64d0dcfc8b1@172.16.2.185
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta8
Date: Tue, 29 Apr 2008 16:30:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1961925072 1961925072 IN IP4 172.16.2.185
s=Asterisk PBX 1.6.0-beta8
c=IN IP4 172.16.2.185
t=0 0
m=audio 13422 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Really destroying SIP dialog '3f2d24a02c67b01212d53b847642f08b@172.16.2.185' Method: INVITE
Retransmitting #1 (no NAT) to 172.16.2.190:5070:
INVITE sip:6136274820@172.16.2.190:5070 SIP/2.0
Via: SIP/2.0/UDP 172.16.2.185:5060;branch=z9hG4bK56ce0e9b;rport
Max-Forwards: 70
From: "Unknown Name" <sip:6137959112@172.16.2.185>;tag=as1edf1b2c
To: <sip:6136274820@172.16.2.190:5070>
Contact: <sip:6137959112@172.16.2.185>
Call-ID: 5dd1a7f12d5e6b201df7f64d0dcfc8b1@172.16.2.185
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta8
Date: Tue, 29 Apr 2008 16:30:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1961925072 1961925072 IN IP4 172.16.2.185
s=Asterisk PBX 1.6.0-beta8
c=IN IP4 172.16.2.185
t=0 0
m=audio 13422 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv