
<--- SIP read from TLS:54.172.60.3:46603 --->
INVITE sips:+12528882095@54.145.2.14;transport=tls SIP/2.0
Record-Route: <sip:54.172.60.3:5061;transport=tls;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
Record-Route: <sip:54.172.60.3:5060;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
From: "+919910199554" <sip:+1919910199554@dev-vsp-trunk-secure.pstn.twilio.com;isup-oli=0;pstn-params=9084818088>;tag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143
To: <sip:+12528882095@54.145.2.14;user=phone>
CSeq: 416691 INVITE
Max-Forwards: 63
P-Asserted-Identity: "+919910199554" <sip:+1919910199554@74.120.92.52:5060>
Diversion: <sip:+12528882095@public-vip.us1.twilio.com>;reason=unconditional
Call-ID: 56da94d3be616b71fcfcfa22c77f8b7e@0.0.0.0
Via: SIP/2.0/TLS 54.172.60.3:5061;branch=z9hG4bKe59e.8271ec21.0
Via: SIP/2.0/UDP 172.18.10.176:5060;rport=5060;received=172.18.10.176;branch=z9hG4bK191bde72-c2f3-43dd-9925-4749c15e1143_6772d868_275-2669896616320982147
Contact: "+919910199554" <sip:+1919910199554@172.18.10.176:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
User-Agent: Twilio Gateway
X-Twilio-AccountSid: AC3177a1945dcd05c575aa44ec29ccc938
Content-Type: application/sdp
X-Twilio-CallSid: CAa5d73178318259a08086ce5cbbfc458e
Content-Length: 321

v=0
o=root 266091639 266091639 IN IP4 54.172.60.185
s=Twilio Media Gateway
c=IN IP4 54.172.60.185
t=0 0
m=audio 16882 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WQaUen04MUZ7GcPBFGvZyheQxZOeW1Xe/T35J1up
<------------->
--- (19 headers 12 lines) ---
Sending to 54.172.60.3:5061 (no NAT)
Sending to 54.172.60.3:5061 (no NAT)
Using INVITE request as basis request - 56da94d3be616b71fcfcfa22c77f8b7e@0.0.0.0
Found peer 'twilio0' for '+1919910199554' from 54.172.60.3:46603
  == 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 - (ulaw|alaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7f763c00cca0 -- Strict RTP learning after remote address set to: 54.172.60.185:16882
Peer audio RTP is at port 54.172.60.185:16882
Looking for +12528882095 in from-twilio (domain 54.145.2.14)
sip_route_dump: route/path hop: <sip:54.172.60.3:5061;transport=tls;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
sip_route_dump: route/path hop: <sip:54.172.60.3:5060;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
RDNIS for this call is +12528882095 (reason unconditional)

<--- Transmitting (no NAT) to 54.172.60.3:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 54.172.60.3:5061;branch=z9hG4bKe59e.8271ec21.0;received=54.172.60.3
Via: SIP/2.0/UDP 172.18.10.176:5060;rport=5060;received=172.18.10.176;branch=z9hG4bK191bde72-c2f3-43dd-9925-4749c15e1143_6772d868_275-2669896616320982147
Record-Route: <sip:54.172.60.3:5061;transport=tls;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
Record-Route: <sip:54.172.60.3:5060;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
From: "+919910199554" <sip:+1919910199554@dev-vsp-trunk-secure.pstn.twilio.com;isup-oli=0;pstn-params=9084818088>;tag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143
To: <sip:+12528882095@54.145.2.14;user=phone>
Call-ID: 56da94d3be616b71fcfcfa22c77f8b7e@0.0.0.0
CSeq: 416691 INVITE
Server: Asterisk PBX 15.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sips:+12528882095@54.145.2.14:5061;transport=tls>
Content-Length: 0


<------------>
    -- Executing [+12528882095@from-twilio:1] Set("SIP/twilio0-00000037", "TrunkProvider=tw") in new stack
    -- Executing [+12528882095@from-twilio:2] Set("SIP/twilio0-00000037", "BillingInfo=CAa5d73178318259a08086ce5cbbfc458e") in new stack
    -- Executing [+12528882095@from-twilio:3] Set("SIP/twilio0-00000037", "REQ-ANSWER-BEFORE-HANGUP=true") in new stack
    -- Executing [+12528882095@from-twilio:4] Stasis("SIP/twilio0-00000037", "app_inst1") in new stack

<--- Transmitting (no NAT) to 54.172.60.3:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 54.172.60.3:5061;branch=z9hG4bKe59e.8271ec21.0;received=54.172.60.3
Via: SIP/2.0/UDP 172.18.10.176:5060;rport=5060;received=172.18.10.176;branch=z9hG4bK191bde72-c2f3-43dd-9925-4749c15e1143_6772d868_275-2669896616320982147
Record-Route: <sip:54.172.60.3:5061;transport=tls;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
Record-Route: <sip:54.172.60.3:5060;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
From: "+919910199554" <sip:+1919910199554@dev-vsp-trunk-secure.pstn.twilio.com;isup-oli=0;pstn-params=9084818088>;tag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143
To: <sip:+12528882095@54.145.2.14;user=phone>;tag=as772addec
Call-ID: 56da94d3be616b71fcfcfa22c77f8b7e@0.0.0.0
CSeq: 416691 INVITE
Server: Asterisk PBX 15.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sips:+12528882095@54.145.2.14:5061;transport=tls>
Content-Length: 0


<------------>
  == Using SIP RTP CoS mark 5
Audio is at 17796
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 54.172.60.0:5061:
INVITE sip:+919811557463@dev-vsp-trunk-secure.pstn.twilio.com SIP/2.0
Via: SIP/2.0/TLS 54.145.2.14:5061;branch=z9hG4bK586ac420
Max-Forwards: 70
From: <sip:1919910199554@54.145.2.14>;tag=as67a6d8ae
To: <sip:+919811557463@dev-vsp-trunk-secure.pstn.twilio.com>
Contact: <sip:1919910199554@54.145.2.14:5061;transport=tls>
Call-ID: 3f3cbe335ee5527749530c8c1d1e1474@54.145.2.14:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.1.2
Date: Thu, 03 May 2018 06:58:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 379

v=0
o=root 503039372 503039372 IN IP4 54.145.2.14
s=Asterisk PBX 15.1.2
c=IN IP4 54.145.2.14
t=0 0
m=audio 17796 RTP/SAVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:6OiD5iUbzyMRFVT4Bio8R85UFrNSXruYX47nvG1e

---
    -- Called +919811557463@dev-vsp-trunk-secure.pstn.twilio.com
  == TLS/SSL ECDH initialized (automatic), faster PFS ciphers enabled
  == TLS/SSL certificate ok

<--- SIP read from TLS:54.172.60.0:5061 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/TLS 54.145.2.14:5061;received=54.145.2.14;rport=57254;branch=z9hG4bK586ac420
From: <sip:1919910199554@54.145.2.14>;tag=as67a6d8ae
To: <sip:+919811557463@dev-vsp-trunk-secure.pstn.twilio.com>
Call-ID: 3f3cbe335ee5527749530c8c1d1e1474@54.145.2.14:5061
CSeq: 102 INVITE
Server: Twilio Gateway
Content-Length: 0

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

<--- SIP read from TLS:54.172.60.0:5061 --->
SIP/2.0 183 Session progress
CSeq: 102 INVITE
Call-ID: 3f3cbe335ee5527749530c8c1d1e1474@54.145.2.14:5061
From: <sip:1919910199554@54.145.2.14>;tag=as67a6d8ae
To: <sip:+919811557463@dev-vsp-trunk-secure.pstn.twilio.com>;tag=76465840_6772d868_fcd83f1a-cbf7-4685-8b24-23fe34eca01a
Via: SIP/2.0/TLS 54.145.2.14:5061;rport=57254;received=54.145.2.14;branch=z9hG4bK586ac420
Record-Route: <sip:54.172.60.0:5060;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
Record-Route: <sip:54.172.60.0:5061;transport=tls;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
Server: Twilio
Contact: <sip:172.18.11.143:5060>
Content-Type: application/sdp
X-Twilio-CallSid: CA809bc193b7de6a7f78e67772d1aef77b
Content-Length: 325

v=0
o=root 2117383904 2117383904 IN IP4 34.203.251.145
s=Twilio Media Gateway
c=IN IP4 34.203.251.145
t=0 0
m=audio 17196 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:jXiBoLhrooTMakDfxJEYM7pEGHJnJnODtU3OVUJi
<------------->
--- (13 headers 12 lines) ---
sip_route_dump: route/path hop: <sip:54.172.60.0:5061;transport=tls;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
sip_route_dump: route/path hop: <sip:54.172.60.0:5060;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
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 - (ulaw|alaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7f763c021d30 -- Strict RTP learning after remote address set to: 34.203.251.145:17196
Peer audio RTP is at port 34.203.251.145:17196
    -- SIP/dev-vsp-trunk-secure.pstn.twilio.com-00000038 is making progress
       > 0x7f763c021d30 -- Strict RTP switching to RTP target address 34.203.251.145:17196 as source
       > 0x7f763c021d30 -- Strict RTP learning complete - Locking on source address 34.203.251.145:17196

<--- SIP read from TLS:54.172.60.0:5061 --->
SIP/2.0 180 Ringing
CSeq: 102 INVITE
Call-ID: 3f3cbe335ee5527749530c8c1d1e1474@54.145.2.14:5061
From: <sip:1919910199554@54.145.2.14>;tag=as67a6d8ae
To: <sip:+919811557463@dev-vsp-trunk-secure.pstn.twilio.com>;tag=76465840_6772d868_fcd83f1a-cbf7-4685-8b24-23fe34eca01a
Via: SIP/2.0/TLS 54.145.2.14:5061;rport=57254;received=54.145.2.14;branch=z9hG4bK586ac420
Record-Route: <sip:54.172.60.0:5060;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
Record-Route: <sip:54.172.60.0:5061;transport=tls;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
Server: Twilio
Contact: <sip:172.18.11.143:5060>
X-Twilio-CallSid: CA809bc193b7de6a7f78e67772d1aef77b
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:54.172.60.0:5061;transport=tls;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
sip_route_dump: route/path hop: <sip:54.172.60.0:5060;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
    -- SIP/dev-vsp-trunk-secure.pstn.twilio.com-00000038 is ringing

<--- SIP read from TLS:54.172.60.0:5061 --->
SIP/2.0 200 OK
CSeq: 102 INVITE
Call-ID: 3f3cbe335ee5527749530c8c1d1e1474@54.145.2.14:5061
From: <sip:1919910199554@54.145.2.14>;tag=as67a6d8ae
To: <sip:+919811557463@dev-vsp-trunk-secure.pstn.twilio.com>;tag=76465840_6772d868_fcd83f1a-cbf7-4685-8b24-23fe34eca01a
Via: SIP/2.0/TLS 54.145.2.14:5061;rport=57254;received=54.145.2.14;branch=z9hG4bK586ac420
Record-Route: <sip:54.172.60.0:5060;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
Record-Route: <sip:54.172.60.0:5061;transport=tls;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
Server: Twilio
Contact: <sip:172.18.11.143:5060>
Content-Type: application/sdp
X-Twilio-CallSid: CA809bc193b7de6a7f78e67772d1aef77b
Content-Length: 325

v=0
o=root 2117383904 2117383904 IN IP4 34.203.251.145
s=Twilio Media Gateway
c=IN IP4 34.203.251.145
t=0 0
m=audio 17196 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:jXiBoLhrooTMakDfxJEYM7pEGHJnJnODtU3OVUJi
<------------->
--- (13 headers 12 lines) ---
sip_route_dump: route/path hop: <sip:54.172.60.0:5061;transport=tls;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
sip_route_dump: route/path hop: <sip:54.172.60.0:5060;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
Transmitting (no NAT) to 172.18.11.143:5060:
ACK sip:172.18.11.143:5060 SIP/2.0
Via: SIP/2.0/TLS 54.145.2.14:5061;branch=z9hG4bK04769dc2
Route: <sip:54.172.60.0:5061;transport=tls;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>,<sip:54.172.60.0:5060;r2=on;lr;ftag=as67a6d8ae;twnat=sip:54.145.2.14:57254>
Max-Forwards: 70
From: <sip:1919910199554@54.145.2.14>;tag=as67a6d8ae
To: <sip:+919811557463@dev-vsp-trunk-secure.pstn.twilio.com>;tag=76465840_6772d868_fcd83f1a-cbf7-4685-8b24-23fe34eca01a
Contact: <sip:1919910199554@54.145.2.14:5061;transport=tls>
Call-ID: 3f3cbe335ee5527749530c8c1d1e1474@54.145.2.14:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.1.2
Content-Length: 0


---
    -- SIP/dev-vsp-trunk-secure.pstn.twilio.com-00000038 answered
    -- Executing [s@app_inst1:1] Set("SIP/dev-vsp-trunk-secure.pstn.twilio.com-00000038", "BillingInfo=") in new stack
    -- Executing [s@app_inst1:2] Stasis("SIP/dev-vsp-trunk-secure.pstn.twilio.com-00000038", "app_inst1") in new stack
Audio is at 14274
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 54.172.60.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 54.172.60.3:5061;branch=z9hG4bKe59e.8271ec21.0;received=54.172.60.3
Via: SIP/2.0/UDP 172.18.10.176:5060;rport=5060;received=172.18.10.176;branch=z9hG4bK191bde72-c2f3-43dd-9925-4749c15e1143_6772d868_275-2669896616320982147
Record-Route: <sip:54.172.60.3:5061;transport=tls;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
Record-Route: <sip:54.172.60.3:5060;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
From: "+919910199554" <sip:+1919910199554@dev-vsp-trunk-secure.pstn.twilio.com;isup-oli=0;pstn-params=9084818088>;tag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143
To: <sip:+12528882095@54.145.2.14;user=phone>;tag=as772addec
Call-ID: 56da94d3be616b71fcfcfa22c77f8b7e@0.0.0.0
CSeq: 416691 INVITE
Server: Asterisk PBX 15.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sips:+12528882095@54.145.2.14:5061;transport=tls>
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 723059078 723059078 IN IP4 54.145.2.14
s=Asterisk PBX 15.1.2
c=IN IP4 54.145.2.14
t=0 0
m=audio 14274 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:jAT6Dwf1V46gQQmZV1ZsCyt/jN4snLZM02H6oynH

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

<--- SIP read from TLS:54.172.60.3:46603 --->
ACK sips:+12528882095@54.145.2.14:5061;transport=tls SIP/2.0
Call-ID: 56da94d3be616b71fcfcfa22c77f8b7e@0.0.0.0
CSeq: 416691 ACK
From: "+919910199554" <sip:+1919910199554@dev-vsp-trunk-secure.pstn.twilio.com;isup-oli=0;pstn-params=9084818088>;tag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143
To: <sip:+12528882095@54.145.2.14;user=phone>;tag=as772addec
Max-Forwards: 69
User-Agent: Twilio
X-Twilio-CallSid: CAa5d73178318259a08086ce5cbbfc458e
Via: SIP/2.0/TLS 54.172.60.3:5061;branch=z9hG4bKe59e.8271ec21.2
Via: SIP/2.0/UDP 172.18.10.176:5060;rport=5060;received=54.211.169.176;branch=z9hG4bK191bde72-c2f3-43dd-9925-4749c15e1143_6772d868_387-2448548136607267129
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
       > 0x7f763c00cca0 -- Strict RTP switching to RTP target address 54.172.60.185:16882 as source
       > 0x7f763c00cca0 -- Strict RTP learning complete - Locking on source address 54.172.60.185:16882
    -- Channel SIP/twilio0-00000037 joined 'simple_bridge' stasis-bridge <1525330721.58>
    -- Channel SIP/dev-vsp-trunk-secure.pstn.twilio.com-00000038 joined 'simple_bridge' stasis-bridge <1525330721.58>

<--- SIP read from TLS:54.172.60.0:5061 --->
BYE sip:1919910199554@54.145.2.14:5061;transport=tls SIP/2.0
CSeq: 1 BYE
From: <sip:+919811557463@dev-vsp-trunk-secure.pstn.twilio.com>;tag=76465840_6772d868_fcd83f1a-cbf7-4685-8b24-23fe34eca01a
To: <sip:1919910199554@54.145.2.14>;tag=as67a6d8ae
Call-ID: 3f3cbe335ee5527749530c8c1d1e1474@54.145.2.14:5061
Max-Forwards: 68
Via: SIP/2.0/TLS 54.172.60.0:5061;branch=z9hG4bKca97.ebbcd6a3.0
Via: SIP/2.0/UDP 172.18.11.143:5060;rport=5060;received=107.23.172.81;branch=z9hG4bKfcd83f1a-cbf7-4685-8b24-23fe34eca01a_6772d868_363-1474503657086317198
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: Twilio Gateway
X-Twilio-CallSid: CA809bc193b7de6a7f78e67772d1aef77b
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 172.18.11.143:5061 (no NAT)
Scheduling destruction of SIP dialog '3f3cbe335ee5527749530c8c1d1e1474@54.145.2.14:5061' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 54.172.60.0:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 54.172.60.0:5061;branch=z9hG4bKca97.ebbcd6a3.0;received=54.172.60.0
Via: SIP/2.0/UDP 172.18.11.143:5060;rport=5060;received=107.23.172.81;branch=z9hG4bKfcd83f1a-cbf7-4685-8b24-23fe34eca01a_6772d868_363-1474503657086317198
From: <sip:+919811557463@dev-vsp-trunk-secure.pstn.twilio.com>;tag=76465840_6772d868_fcd83f1a-cbf7-4685-8b24-23fe34eca01a
To: <sip:1919910199554@54.145.2.14>;tag=as67a6d8ae
Call-ID: 3f3cbe335ee5527749530c8c1d1e1474@54.145.2.14:5061
CSeq: 1 BYE
Server: Asterisk PBX 15.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/dev-vsp-trunk-secure.pstn.twilio.com-00000038 left 'simple_bridge' stasis-bridge <1525330721.58>
    -- Channel SIP/twilio0-00000037 left 'simple_bridge' stasis-bridge <1525330721.58>
Scheduling destruction of SIP dialog '56da94d3be616b71fcfcfa22c77f8b7e@0.0.0.0' in 32000 ms (Method: ACK)
Reliably Transmitting (no NAT) to 54.172.60.3:5061:
BYE sip:+1919910199554@172.18.10.176:5060;transport=udp SIP/2.0
Via: SIP/2.0/TLS 54.145.2.14:5061;branch=z9hG4bK2776d569
Route: <sip:54.172.60.3:5061;transport=tls;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>,<sip:54.172.60.3:5060;r2=on;lr;ftag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143>
Max-Forwards: 70
From: <sip:+12528882095@54.145.2.14;user=phone>;tag=as772addec
To: "+919910199554" <sip:+1919910199554@dev-vsp-trunk-secure.pstn.twilio.com;isup-oli=0;pstn-params=9084818088>;tag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143
Call-ID: 56da94d3be616b71fcfcfa22c77f8b7e@0.0.0.0
CSeq: 102 BYE
User-Agent: Asterisk PBX 15.1.2
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from TLS:54.172.60.3:46603 --->
SIP/2.0 200 OK
CSeq: 102 BYE
Call-ID: 56da94d3be616b71fcfcfa22c77f8b7e@0.0.0.0
From: <sip:+12528882095@54.145.2.14;user=phone>;tag=as772addec
To: "+919910199554" <sip:+1919910199554@dev-vsp-trunk-secure.pstn.twilio.com;isup-oli=0;pstn-params=9084818088>;tag=89579822_6772d868_191bde72-c2f3-43dd-9925-4749c15e1143
Via: SIP/2.0/TLS 54.145.2.14:5061;rport=5061;received=54.145.2.14;branch=z9hG4bK2776d569
Server: Twilio
X-Twilio-CallSid: CAa5d73178318259a08086ce5cbbfc458e
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '56da94d3be616b71fcfcfa22c77f8b7e@0.0.0.0' Method: ACK
Really destroying SIP dialog '3f3cbe335ee5527749530c8c1d1e1474@54.145.2.14:5061' Method: BYE