== Using SIP RTP CoS mark 5
    -- Executing [12012156850@inbound:1] Answer("SIP/flowroute-00000027", "") in new stack
    -- Auto fallthrough, channel 'SIP/flowroute-00000027' status is 'UNKNOWN'
asterisknow*CLI> sip set debug on
SIP Debugging enabled
asterisknow*CLI>

<--- SIP read from UDP:216.115.69.144:5060 --->
INVITE sip:12012156850@192.168.0.236:5060 SIP/2.0
Record-Route: <sip:216.115.69.144;lr>
Record-Route: <sip:216.115.69.132;lr>
To: <sip:+12012156850@flowroute.com>
From: "Unavailable " <sip:UNAVAILABLE@flowroute.com>;tag=gK0937d977
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK7d6c.ee081fb9c004df8eb554ff8bc523bc76.0
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK7d6c.05c02c80dc6e14a98aeb4c0f781e1d34.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK7d6c.02d9a615ffed650339c953cd539e6171.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK09B62c535a46c5cb738
Call-ID: 856237267_119137348@4.55.17.35
CSeq: 1978 INVITE
Max-Forwards: 66
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: "Unavailable" <sip:Restricted@4.55.17.35:5060>
Content-Length: 225
Content-Type: application/sdp
Diversion: <sip:+12012156850@flowroute.com>;reason=unknown;screen=yes;privacy=off
Diversion: <sip:+12012156850@flowroute.com>;reason=unknown;screen=yes;privacy=off
P-Asserted-Identity: "Unavailable " <sip:UNAVAILABLE@flowroute.com>

v=0
o=- 13304 8470 IN IP4 4.55.17.2
s=-
c=IN IP4 4.55.17.2
t=0 0
m=audio 22594 RTP/AVP 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20
<------------->
--- (19 headers 12 lines) ---
Sending to 216.115.69.144:5060 (no NAT)
Using INVITE request as basis request - 856237267_119137348@4.55.17.35
Found peer 'flowroute' for 'UNAVAILABLE' from 216.115.69.144:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008010e (gsm|ulaw|alaw|g729|h263|testlaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 4.55.17.2:22594
Looking for 12012156850 in inbound (domain 192.168.0.236:5060)
list_route: hop: <sip:216.115.69.144;lr>
list_route: hop: <sip:216.115.69.132;lr>
RDNIS for this call is +12012156850 (reason unknown)

<--- Transmitting (NAT) to 216.115.69.144:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK7d6c.ee081fb9c004df8eb554ff8bc523bc76.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK7d6c.05c02c80dc6e14a98aeb4c0f781e1d34.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK7d6c.02d9a615ffed650339c953cd539e6171.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK09B62c535a46c5cb738
Record-Route: <sip:216.115.69.144;lr>
Record-Route: <sip:216.115.69.132;lr>
From: "Unavailable " <sip:UNAVAILABLE@flowroute.com>;tag=gK0937d977
To: <sip:+12012156850@flowroute.com>
Call-ID: 856237267_119137348@4.55.17.35
CSeq: 1978 INVITE
Server: Asterisk PBX 1.8.4.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:12012156850@192.168.0.236:5060>
Content-Length: 0


<------------>
    -- Executing [12012156850@inbound:1] Answer("SIP/flowroute-00000028", "") in new stack
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 216.115.69.144:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK7d6c.ee081fb9c004df8eb554ff8bc523bc76.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK7d6c.05c02c80dc6e14a98aeb4c0f781e1d34.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK7d6c.02d9a615ffed650339c953cd539e6171.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK09B62c535a46c5cb738
Record-Route: <sip:216.115.69.144;lr>
Record-Route: <sip:216.115.69.132;lr>
From: "Unavailable " <sip:UNAVAILABLE@flowroute.com>;tag=gK0937d977
To: <sip:+12012156850@flowroute.com>;tag=as0d37d99e
Call-ID: 856237267_119137348@4.55.17.35
CSeq: 1978 INVITE
Server: Asterisk PBX 1.8.4.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:12012156850@192.168.0.236:5060>
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 360701665 360701665 IN IP4 192.168.0.236
s=Asterisk PBX 1.8.4.4
c=IN IP4 192.168.0.236
t=0 0
m=audio 13078 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:216.115.69.144:5060 --->
ACK sip:12012156850@121.135.82.142:5060 SIP/2.0
Record-Route: <sip:216.115.69.144;lr>
Record-Route: <sip:216.115.69.132;lr>
From: "Unavailable " <sip:UNAVAILABLE@flowroute.com>;tag=gK0937d977
To: <sip:+12012156850@flowroute.com>;tag=as0d37d99e
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK7d6c.de7cd3ab194734fc924eab1c25eb157a.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK7d6c.d7b1972378708ab874465150838e1029.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK09B62c603a0daad83ad
Call-ID: 856237267_119137348@4.55.17.35
CSeq: 1978 ACK
Max-Forwards: 68
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
    -- Auto fallthrough, channel 'SIP/flowroute-00000028' status is 'UNKNOWN'
Scheduling destruction of SIP dialog '856237267_119137348@4.55.17.35' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:216.115.69.144;lr> for address/port to send to
set_destination: set destination to 216.115.69.144:5060
Reliably Transmitting (NAT) to 216.115.69.144:5060:
BYE sip:Restricted@4.55.17.35:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.236:5060;branch=z9hG4bK5fbf811a;rport
Route: <sip:216.115.69.144;lr>,<sip:216.115.69.132;lr>
Max-Forwards: 70
From: <sip:+12012156850@flowroute.com>;tag=as0d37d99e
To: "Unavailable " <sip:UNAVAILABLE@flowroute.com>;tag=gK0937d977
Call-ID: 856237267_119137348@4.55.17.35
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.4.4
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---

<--- SIP read from UDP:216.115.69.144:5060 --->
SIP/2.0 200 OK
From: <sip:+12012156850@flowroute.com>;tag=as0d37d99e
To: "Unavailable " <sip:UNAVAILABLE@flowroute.com>;tag=gK0937d977
Via: SIP/2.0/UDP 192.168.0.236:5060;received=121.135.82.142;branch=z9hG4bK5fbf811a;rport=5060
Call-ID: 856237267_119137348@4.55.17.35
CSeq: 102 BYE
Record-Route: <sip:216.115.69.132:5060;lr>
Record-Route: <sip:216.115.69.144:5060;lr>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '856237267_119137348@4.55.17.35' Method: ACK
Really destroying SIP dialog '537120d7-31b353b6-c09c105@70.167.153.136' Method: OPTIONS

<--- SIP read from UDP:216.115.69.144:5060 --->
OPTIONS sip:121.135.82.142:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:216.115.69.144;lr>
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKb676.e574c80a10f23f3d053034661a7aff8e.0
Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
Route: <sip:216.115.69.144;lr;received="sip:121.135.82.142:5060">
From: sip:ping@invalid;tag=99bb8871
To: sip:121.135.82.142:5060
Call-ID: 4a13cc25-600ab215-7fc4a62@216.115.69.131
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Looking for  in default (domain 121.135.82.142:5060)

<--- Transmitting (no NAT) to 216.115.69.144:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKb676.e574c80a10f23f3d053034661a7aff8e.0;received=216.115.69.144
Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
From: sip:ping@invalid;tag=99bb8871
To: sip:121.135.82.142:5060;tag=as7a15ca0e
Call-ID: 4a13cc25-600ab215-7fc4a62@216.115.69.131
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.4.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4a13cc25-600ab215-7fc4a62@216.115.69.131' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:216.115.69.144:5060 --->
OPTIONS sip:121.135.82.142:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:216.115.69.144;lr>
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKcdd5.af2d2d515236ed99b85a2fb3f7086b6d.0
Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
Route: <sip:216.115.69.144;lr;received="sip:121.135.82.142:5060">
From: sip:ping@invalid;tag=7c9722d2
To: sip:121.135.82.142:5060
Call-ID: 537120d7-ca5453b6-349c105@70.167.153.136
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Looking for  in default (domain 121.135.82.142:5060)

<--- Transmitting (no NAT) to 216.115.69.144:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKcdd5.af2d2d515236ed99b85a2fb3f7086b6d.0;received=216.115.69.144
Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
From: sip:ping@invalid;tag=7c9722d2
To: sip:121.135.82.142:5060;tag=as3983474a
Call-ID: 537120d7-ca5453b6-349c105@70.167.153.136
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.4.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '537120d7-ca5453b6-349c105@70.167.153.136' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '4a13cc25-600ab215-7fc4a62@216.115.69.131' Method: OPTIONS
Really destroying SIP dialog '537120d7-ca5453b6-349c105@70.167.153.136' Method: OPTIONS

<--- SIP read from UDP:216.115.69.144:5060 --->
OPTIONS sip:121.135.82.142:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:216.115.69.144;lr>
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK3c0e.03dde99bbc74230416a93afbe0bc31a3.0
Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
Route: <sip:216.115.69.144;lr;received="sip:121.135.82.142:5060">
From: sip:ping@invalid;tag=b26c8871
To: sip:121.135.82.142:5060
Call-ID: 4a13cc25-89aab215-e2d4a62@216.115.69.131
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Looking for  in default (domain 121.135.82.142:5060)

<--- Transmitting (no NAT) to 216.115.69.144:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK3c0e.03dde99bbc74230416a93afbe0bc31a3.0;received=216.115.69.144
Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
From: sip:ping@invalid;tag=b26c8871
To: sip:121.135.82.142:5060;tag=as0cee7eab
Call-ID: 4a13cc25-89aab215-e2d4a62@216.115.69.131
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.4.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4a13cc25-89aab215-e2d4a62@216.115.69.131' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:216.115.69.144:5060 --->
OPTIONS sip:121.135.82.142:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:216.115.69.144;lr>
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK665d.ac09005ef697e9c952b939904dcbc4a9.0
Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
Route: <sip:216.115.69.144;lr;received="sip:121.135.82.142:5060">
From: sip:ping@invalid;tag=b54822d2
To: sip:121.135.82.142:5060
Call-ID: 537120d7-040553b6-a79c105@70.167.153.136
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Looking for  in default (domain 121.135.82.142:5060)

<--- Transmitting (no NAT) to 216.115.69.144:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK665d.ac09005ef697e9c952b939904dcbc4a9.0;received=216.115.69.144
Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
From: sip:ping@invalid;tag=b54822d2
To: sip:121.135.82.142:5060;tag=as150d015f
Call-ID: 537120d7-040553b6-a79c105@70.167.153.136
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.4.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '537120d7-040553b6-a79c105@70.167.153.136' in 3200