<--- SIP read from UDP:32.163.38.200:34537 --->
INVITE sip:8005551212@voip.routed.com SIP/2.0
Via: SIP/2.0/UDP 32.163.38.200:34537;rport;branch=z9hG4bKPjk.4m36nfM9t2kI9wZN40bHrf63CPVyfF
Max-Forwards: 70
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com
Contact: <sip:201@32.163.38.200:34537>
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5852 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Bria Android 2.0.3
Content-Type: application/sdp
Content-Length: 210

v=0
o=- 3553718701 3553718701 IN IP4 32.163.38.200
s=cpc_med
c=IN IP4 32.163.38.200
t=0 0
m=audio 14008 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 10 lines) ---
Sending to 32.163.38.200:34537 (NAT)
Using INVITE request as basis request - P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
Found peer '201' for '201' from 32.163.38.200:34537

<--- Reliably Transmitting (NAT) to 32.163.38.200:34537 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 32.163.38.200:34537;branch=z9hG4bKPjk.4m36nfM9t2kI9wZN40bHrf63CPVyfF;received=32.163.38.200;rport=34537
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com;tag=as4f1b2a83
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5852 INVITE
Server: Star2Star StarBox astlinux-s2s-4918-via
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="routed.com", nonce="7d36b6d9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'P.8elr.rWbZilkZS0YbzU2X.04-ooP8v' in 13248 ms (Method: INVITE)
Really destroying SIP dialog '091bd2f97ce33e7574ebeb2958136ac2@[2001:470:7:262::2]' Method: REGISTER
Retransmitting #1 (NAT) to 32.163.38.200:34537:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 32.163.38.200:34537;branch=z9hG4bKPjk.4m36nfM9t2kI9wZN40bHrf63CPVyfF;received=32.163.38.200;rport=34537
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com;tag=as4f1b2a83
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5852 INVITE
Server: Star2Star StarBox astlinux-s2s-4918-via
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="routed.com", nonce="7d36b6d9"
Content-Length: 0


---

<--- SIP read from UDP:32.163.38.200:34537 --->
ACK sip:8005551212@voip.routed.com SIP/2.0
Via: SIP/2.0/UDP 32.163.38.200:34537;rport;branch=z9hG4bKPjk.4m36nfM9t2kI9wZN40bHrf63CPVyfF
Max-Forwards: 70
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com;tag=as4f1b2a83
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5852 ACK
Content-Length: 0

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

<--- SIP read from UDP:32.163.38.200:34537 --->
INVITE sip:8005551212@voip.routed.com SIP/2.0
Via: SIP/2.0/UDP 32.163.38.200:34537;rport;branch=z9hG4bKPjmDLIsbqEF3nvgQB-eYzY3VPWl3Eq.2DH
Max-Forwards: 70
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com
Contact: <sip:201@32.163.38.200:34537>
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5853 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Bria Android 2.0.3
Authorization: Digest username="201", realm="routed.com", nonce="7d36b6d9", uri="sip:8005551212@voip.routed.com", response="9be4dff3993259492b65476091a816ea", algorithm=MD5
Content-Type: application/sdp
Content-Length: 210

v=0
o=- 3553718701 3553718701 IN IP4 32.163.38.200
s=cpc_med
c=IN IP4 32.163.38.200
t=0 0
m=audio 14008 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (16 headers 10 lines) ---
Sending to 32.163.38.200:34537 (NAT)
Using INVITE request as basis request - P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
Found peer '201' for '201' from 32.163.38.200:34537
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x1106 (gsm|ulaw|g729|g722), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 32.163.38.200:14008
Looking for 8005551212 in from-internal (domain voip.routed.com)
list_route: hop: <sip:201@32.163.38.200:34537>

<--- Transmitting (NAT) to 32.163.38.200:34537 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 32.163.38.200:34537;branch=z9hG4bKPjmDLIsbqEF3nvgQB-eYzY3VPWl3Eq.2DH;received=32.163.38.200;rport=34537
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5853 INVITE
Server: Star2Star StarBox astlinux-s2s-4918-via
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:8005551212@71.180.124.149:5060>
Content-Length: 0


<------------>
    -- Executing [8005551212@from-internal:1] Set("SIP/201-000000e1", "__DYNAMIC_FEATURES=automon") in new stack
    -- Executing [8005551212@from-internal:2] Set("SIP/201-000000e1", "__MONITOR_EXEC=/usr/bin/soxmix.sh") in new stack
    -- Executing [8005551212@from-internal:3] MixMonitor("SIP/201-000000e1", "/mnt/kd/monitor/rec-201-8005551212-1344729903.wav") in new stack
  == Begin MixMonitor Recording SIP/201-000000e1
    -- Executing [8005551212@from-internal:4] Macro("SIP/201-000000e1", "dialstacked,,8005551212,60,wW") in new stack
    -- Executing [s@macro-dialstacked:1] SIPAddHeader("SIP/201-000000e1", "X-s2s-region: 1") in new stack
    -- Executing [s@macro-dialstacked:2] Set("SIP/201-000000e1", "PREFIX=+1") in new stack
    -- Executing [s@macro-dialstacked:3] Set("SIP/201-000000e1", "CALLERID(num)=201") in new stack
    -- Executing [s@macro-dialstacked:4] Set("SIP/201-000000e1", "LENGTH=3") in new stack
    -- Executing [s@macro-dialstacked:5] Set("SIP/201-000000e1", "IGNORECIDLENGTH=0") in new stack
    -- Executing [s@macro-dialstacked:6] GosubIf("SIP/201-000000e1", "0?8") in new stack
    -- Executing [s@macro-dialstacked:7] Set("SIP/201-000000e1", "CALLERID(num)=9412966982") in new stack
    -- Executing [s@macro-dialstacked:8] Macro("SIP/201-000000e1", "dials2s,+18005551212,60,wW") in new stack
    -- Executing [s@macro-dials2s:1] NoOp("SIP/201-000000e1", ""Dialing s2s with +18005551212, 60, wW"") in new stack
    -- Executing [s@macro-dials2s:2] Gosub("SIP/201-000000e1", "dial-sip2-atl,1") in new stack
    -- Executing [dial-sip2-atl@macro-dials2s:1] Wait("SIP/201-000000e1", ".25") in new stack

<--- SIP read from UDP:32.163.38.200:34537 --->
ACK sip:8005551212@voip.routed.com SIP/2.0
Via: SIP/2.0/UDP 32.163.38.200:34537;rport;branch=z9hG4bKPjk.4m36nfM9t2kI9wZN40bHrf63CPVyfF
Max-Forwards: 70
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com;tag=as4f1b2a83
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5852 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- Executing [dial-sip2-atl@macro-dials2s:2] Dial("SIP/201-000000e1", "SIP/S2S_TRUNK-atl/+18005551212,60,wW") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11448
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 199.15.180.2:5060:
INVITE sip:+18005551212@199.15.180.2 SIP/2.0
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK12c70a3e;rport
Max-Forwards: 70
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>
Contact: <sip:9412966982@71.180.124.149:5060>
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 102 INVITE
User-Agent: Star2Star StarBox astlinux-s2s-4918-via
Date: Sun, 12 Aug 2012 00:05:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-s2s-region: 1
Remote-Party-ID: "James SoftPhone" <sip:9412966982@71.180.124.149>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 510908894 510908894 IN IP4 71.180.124.149
s=Asterisk PBX 1.8.14.1
c=IN IP4 71.180.124.149
t=0 0
m=audio 11448 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

---
    -- Called SIP/S2S_TRUNK-atl/+18005551212

<--- SIP read from UDP:199.15.180.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK12c70a3e;rport=5060
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:199.15.180.2:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK12c70a3e;rport=5060
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>;tag=984f7d53c9f70c9ab287a3ba0dc4abf7.e125
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="star2star.com", nonce="UCb0XFAm8zBBeLJgOIiKAhDPvuaZs17e"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 199.15.180.2:5060:
ACK sip:+18005551212@199.15.180.2 SIP/2.0
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK12c70a3e;rport
Max-Forwards: 70
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>;tag=984f7d53c9f70c9ab287a3ba0dc4abf7.e125
Contact: <sip:9412966982@71.180.124.149:5060>
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 102 ACK
User-Agent: Star2Star StarBox astlinux-s2s-4918-via
Content-Length: 0


---
Audio is at 11448
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 199.15.180.2:5060:
INVITE sip:+18005551212@199.15.180.2 SIP/2.0
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK10556e92;rport
Max-Forwards: 70
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>
Contact: <sip:9412966982@71.180.124.149:5060>
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 103 INVITE
User-Agent: Star2Star StarBox astlinux-s2s-4918-via
Proxy-Authorization: Digest username="starbox_143", realm="star2star.com", algorithm=MD5, uri="sip:+18005551212@199.15.180.2", nonce="UCb0XFAm8zBBeLJgOIiKAhDPvuaZs17e", response="17f5e10afe2b174045f347834057fa40"
Date: Sun, 12 Aug 2012 00:05:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-s2s-region: 1
Remote-Party-ID: "James SoftPhone" <sip:9412966982@71.180.124.149>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 510908894 510908895 IN IP4 71.180.124.149
s=Asterisk PBX 1.8.14.1
c=IN IP4 71.180.124.149
t=0 0
m=audio 11448 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:199.15.180.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK10556e92;rport=5060
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 103 INVITE
Content-Length: 0

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

<--- SIP read from UDP:199.15.180.2:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK10556e92;rport=5060
Record-Route: <sip:199.15.180.2;lr=on;ftag=as7e6482a4;did=5a7.8620edd1>
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>;tag=Br5Q661KcaQ0j
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 103 INVITE
Contact: <sip:+18005551212@199.15.180.1:5080;transport=udp>
User-Agent: Star2Star Media
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 219

v=0
o=Sonus_UAC 21391 1452 IN IP4 4.55.6.163
s=SIP Media Capabilities
c=IN IP4 4.55.6.130
t=0 0
m=audio 16618 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
--- (16 headers 10 lines) ---
list_route: hop: <sip:199.15.180.2;lr=on;ftag=as7e6482a4;did=5a7.8620edd1>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 4.55.6.130:16618
<< [ TYPE: Control (4) SUBCLASS: Unknown control '14' (14) ] [SIP/S2S_TRUNK-atl-000000e2]
    -- SIP/S2S_TRUNK-atl-000000e2 is making progress passing it to SIP/201-000000e1
Audio is at 17122
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 32.163.38.200:34537 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 32.163.38.200:34537;branch=z9hG4bKPjmDLIsbqEF3nvgQB-eYzY3VPWl3Eq.2DH;received=32.163.38.200;rport=34537
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com;tag=as3aa1d699
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5853 INVITE
Server: Star2Star StarBox astlinux-s2s-4918-via
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:8005551212@71.180.124.149:5060>
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 2105266401 2105266401 IN IP4 71.180.124.149
s=Asterisk PBX 1.8.14.1
c=IN IP4 71.180.124.149
t=0 0
m=audio 17122 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:199.15.180.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK10556e92;rport=5060
Record-Route: <sip:199.15.180.2;lr=on;ftag=as7e6482a4;did=5a7.8620edd1>
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>;tag=Br5Q661KcaQ0j
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 103 INVITE
Contact: <sip:+18005551212@199.15.180.1:5080;transport=udp>
User-Agent: Star2Star Media
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 219

v=0
o=Sonus_UAC 21391 1452 IN IP4 4.55.6.163
s=SIP Media Capabilities
c=IN IP4 4.55.6.130
t=0 0
m=audio 16618 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
--- (15 headers 10 lines) ---
list_route: hop: <sip:199.15.180.2;lr=on;ftag=as7e6482a4;did=5a7.8620edd1>
set_destination: Parsing <sip:199.15.180.2;lr=on;ftag=as7e6482a4;did=5a7.8620edd1> for address/port to send to
set_destination: set destination to 199.15.180.2:5060
Transmitting (NAT) to 199.15.180.2:5060:
ACK sip:+18005551212@199.15.180.1:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK0536003e;rport
Route: <sip:199.15.180.2;lr=on;ftag=as7e6482a4;did=5a7.8620edd1>
Max-Forwards: 70
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>;tag=Br5Q661KcaQ0j
Contact: <sip:9412966982@71.180.124.149:5060>
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 103 ACK
User-Agent: Star2Star StarBox astlinux-s2s-4918-via
Content-Length: 0


---
<< [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/S2S_TRUNK-atl-000000e2]
    -- SIP/S2S_TRUNK-atl-000000e2 answered SIP/201-000000e1
Audio is at 17122
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 32.163.38.200:34537 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 32.163.38.200:34537;branch=z9hG4bKPjmDLIsbqEF3nvgQB-eYzY3VPWl3Eq.2DH;received=32.163.38.200;rport=34537
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com;tag=as3aa1d699
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5853 INVITE
Server: Star2Star StarBox astlinux-s2s-4918-via
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:8005551212@71.180.124.149:5060>
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 2105266401 2105266402 IN IP4 71.180.124.149
s=Asterisk PBX 1.8.14.1
c=IN IP4 71.180.124.149
t=0 0
m=audio 17122 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:32.163.38.200:34537 --->
ACK sip:8005551212@71.180.124.149:5060 SIP/2.0
Via: SIP/2.0/UDP 32.163.38.200:34537;rport;branch=z9hG4bKPjrGzB3cg3ICFYo2uElXu60xXElLNVlYF1
Max-Forwards: 70
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com;tag=as3aa1d699
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5853 ACK
Content-Length: 0

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

<--- SIP read from UDP:32.163.38.200:34537 --->

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

<--- SIP read from UDP:32.163.38.200:34537 --->
BYE sip:8005551212@71.180.124.149:5060 SIP/2.0
Via: SIP/2.0/UDP 32.163.38.200:34537;rport;branch=z9hG4bKPjE-ijt44SuEZi28BRXNNj5XIVwcFZaRlE
Max-Forwards: 70
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com;tag=as3aa1d699
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5854 BYE
User-Agent: Bria Android 2.0.3
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 32.163.38.200:34537 (NAT)
Scheduling destruction of SIP dialog 'P.8elr.rWbZilkZS0YbzU2X.04-ooP8v' in 13248 ms (Method: BYE)

<--- Transmitting (NAT) to 32.163.38.200:34537 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 32.163.38.200:34537;branch=z9hG4bKPjE-ijt44SuEZi28BRXNNj5XIVwcFZaRlE;received=32.163.38.200;rport=34537
From: <sip:201@voip.routed.com>;tag=lP46f45DG.1M6u2TlynO04Ssx2zjiTdq
To: sip:8005551212@voip.routed.com;tag=as3aa1d699
Call-ID: P.8elr.rWbZilkZS0YbzU2X.04-ooP8v
CSeq: 5854 BYE
Server: Star2Star StarBox astlinux-s2s-4918-via
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
<< [ HANGUP (NULL) ] [SIP/201-000000e1]
Scheduling destruction of SIP dialog '0027067c2230d83a3e805ae843bececb@71.180.124.149:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:199.15.180.2;lr=on;ftag=as7e6482a4;did=5a7.8620edd1> for address/port to send to
set_destination: set destination to 199.15.180.2:5060
Reliably Transmitting (NAT) to 199.15.180.2:5060:
BYE sip:+18005551212@199.15.180.1:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK2839cff6;rport
Route: <sip:199.15.180.2;lr=on;ftag=as7e6482a4;did=5a7.8620edd1>
Max-Forwards: 70
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>;tag=Br5Q661KcaQ0j
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 104 BYE
User-Agent: Star2Star StarBox astlinux-s2s-4918-via
Proxy-Authorization: Digest username="starbox_143", realm="star2star.com", algorithm=MD5, uri="sip:+18005551212@199.15.180.1:5080", nonce="UCb0XFAm8zBBeLJgOIiKAhDPvuaZs17e", response="f4a49fab1fcba519a8af3b905ac344d9"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dials2s, dial-sip2-atl, 2) exited non-zero on 'SIP/201-000000e1' in macro 'dials2s'
  == Spawn extension (macro-dialstacked, s, 8) exited non-zero on 'SIP/201-000000e1' in macro 'dialstacked'
  == Spawn extension (from-internal, 8005551212, 4) exited non-zero on 'SIP/201-000000e1'
  == End MixMonitor Recording SIP/201-000000e1

<--- SIP read from UDP:199.15.180.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 71.180.124.149:5060;branch=z9hG4bK2839cff6;rport=5060
From: "James SoftPhone" <sip:9412966982@71.180.124.149>;tag=as7e6482a4
To: <sip:+18005551212@199.15.180.2>;tag=Br5Q661KcaQ0j
Call-ID: 0027067c2230d83a3e805ae843bececb@71.180.124.149:5060
CSeq: 104 BYE
User-Agent: Star2Star Media
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0027067c2230d83a3e805ae843bececb@71.180.124.149:5060' Method: INVITE