Asterisk 13.1-cert2, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.1-cert2 currently running on ubuntu (pid = 1105)

<--- SIP read from UDP:10.16.1.200:5060 --->
PUBLISH sip:221@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPj55192c787766438ba49b07f73b5bb334
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=cdae0100fe5e4471a1a1a1287256deff
To: "221" <sip:221@10.16.1.220>
Call-ID: 46fab42864494c90a713701e2e082bc4
CSeq: 32164 PUBLISH
Event: presence
User-Agent: MicroSIP/3.10.9
Content-Type: application/pidf+xml
Content-Length: 549

<?xml version="1.0" encoding="UTF-8"?>
<presence entity="sip:221@10.16.1.220" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"> <tuple id="pjec301114dee6476bbac4733118ca4a3b"> <status> <basic>open</basic> </status> <timestamp>2016-01-15T16:52:11.144Z</timestamp> <note>Away</note> </tuple> <dm:person id="pj720a82fc407c4f15a78ab6bc1da3b6eb"> <rpid:activities> <rpid:away /> </rpid:activities> <dm:note>Away</dm:note> </dm:person>
</presence>
<------------->
--- (11 headers 3 lines) ---
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj55192c787766438ba49b07f73b5bb334;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=cdae0100fe5e4471a1a1a1287256deff
To: "221" <sip:221@10.16.1.220>;tag=as3c2c4a36
Call-ID: 46fab42864494c90a713701e2e082bc4
CSeq: 32164 PUBLISH
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '46fab42864494c90a713701e2e082bc4' Method: PUBLISH

<--- SIP read from UDP:10.16.1.146:54896 --->


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

<--- SIP read from UDP:10.16.1.200:5060 --->

<------------->
Reliably Transmitting (no NAT) to 10.16.1.146:54896:
OPTIONS sip:220@10.16.1.146:54896;line=ecbf1a5cf131885 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK6e41938b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.16.1.220>;tag=as49d1cc41
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>
Contact: <sip:asterisk@10.16.1.220:5060>
Call-ID: 1a1623ef56e4219461e3982610ec3d63@10.16.1.220:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1-cert2
Date: Fri, 15 Jan 2016 15:52:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.16.1.146:54896 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK6e41938b
From: "asterisk" <sip:asterisk@10.16.1.220>;tag=as49d1cc41
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>;tag=1977816965
Call-ID: 1a1623ef56e4219461e3982610ec3d63@10.16.1.220:5060
CSeq: 102 OPTIONS
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Accept: application/sdp
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Supported: 100rel, replaces
Content-Length: 383

v=0
o=amsip 0 0 IN IP4 10.16.1.146
s=talk
c=IN IP4 10.16.1.146
t=0 0
m=audio 0 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 118 117 116 115
b=AS:128
a=rtpmap:118 VP8/90000
a=rtpmap:117 H264/90000
a=fmtp:117 profile-level-id=42800c; packetization-mode=1
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000
<------------->
--- (12 headers 16 lines) ---
Really destroying SIP dialog '1a1623ef56e4219461e3982610ec3d63@10.16.1.220:5060' Method: OPTIONS

<--- SIP read from UDP:10.16.1.146:54896 --->


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

<--- SIP read from UDP:10.16.1.200:5060 --->
REGISTER sip:10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPjf81d67fad82045d3ab87d64d654759d9
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=9b7f9e7f4985407fa395009623993043
To: "221" <sip:221@10.16.1.220>
Call-ID: 1d3a56c2ca1b41d2adc4597bc2faf653
CSeq: 45333 REGISTER
User-Agent: MicroSIP/3.10.9
Contact: "221" <sip:221@10.16.1.200:5060;ob>
Expires: 0
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 10.16.1.200:5060 (no NAT)
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPjf81d67fad82045d3ab87d64d654759d9;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=9b7f9e7f4985407fa395009623993043
To: "221" <sip:221@10.16.1.220>;tag=as74a984a1
Call-ID: 1d3a56c2ca1b41d2adc4597bc2faf653
CSeq: 45333 REGISTER
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3fee7dec"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1d3a56c2ca1b41d2adc4597bc2faf653' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.16.1.200:5060 --->
REGISTER sip:10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPje3a836b5f0ad487c924427935f2bf0ba
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=9b7f9e7f4985407fa395009623993043
To: "221" <sip:221@10.16.1.220>
Call-ID: 1d3a56c2ca1b41d2adc4597bc2faf653
CSeq: 45334 REGISTER
User-Agent: MicroSIP/3.10.9
Contact: "221" <sip:221@10.16.1.200:5060;ob>
Expires: 0
Authorization: Digest username="221", realm="asterisk", nonce="3fee7dec", uri="sip:10.16.1.220", response="3c6e94bec69a95d72eb2fbc10487a264", algorithm=MD5
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPje3a836b5f0ad487c924427935f2bf0ba;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=9b7f9e7f4985407fa395009623993043
To: "221" <sip:221@10.16.1.220>;tag=as74a984a1
Call-ID: 1d3a56c2ca1b41d2adc4597bc2faf653
CSeq: 45334 REGISTER
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 0
Date: Fri, 15 Jan 2016 15:52:34 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1d3a56c2ca1b41d2adc4597bc2faf653' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.16.1.200:5060 --->
REGISTER sip:10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPjfaf23dbad3f1474a9f69908a42b0907c
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=079aa88f1ced41d5a617d5eb59d8a872
To: "221" <sip:221@10.16.1.220>
Call-ID: e752fc90a1ee4b4aabe1dd5dabad5cc6
CSeq: 40145 REGISTER
User-Agent: MicroSIP/3.10.9
Contact: "221" <sip:221@10.16.1.200:5060;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 10.16.1.200:5060 (no NAT)
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPjfaf23dbad3f1474a9f69908a42b0907c;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=079aa88f1ced41d5a617d5eb59d8a872
To: "221" <sip:221@10.16.1.220>;tag=as166b293d
Call-ID: e752fc90a1ee4b4aabe1dd5dabad5cc6
CSeq: 40145 REGISTER
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3dd5a022"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e752fc90a1ee4b4aabe1dd5dabad5cc6' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.16.1.200:5060 --->
REGISTER sip:10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPj62f5519ca87246ea812a700ac980cc85
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=079aa88f1ced41d5a617d5eb59d8a872
To: "221" <sip:221@10.16.1.220>
Call-ID: e752fc90a1ee4b4aabe1dd5dabad5cc6
CSeq: 40146 REGISTER
User-Agent: MicroSIP/3.10.9
Contact: "221" <sip:221@10.16.1.200:5060;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="221", realm="asterisk", nonce="3dd5a022", uri="sip:10.16.1.220", response="d60ebb994e2eaa2e14337ef3ace22462", algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj62f5519ca87246ea812a700ac980cc85;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=079aa88f1ced41d5a617d5eb59d8a872
To: "221" <sip:221@10.16.1.220>;tag=as166b293d
Call-ID: e752fc90a1ee4b4aabe1dd5dabad5cc6
CSeq: 40146 REGISTER
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: <sip:221@10.16.1.200:5060;ob>;expires=300
Date: Fri, 15 Jan 2016 15:52:36 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e752fc90a1ee4b4aabe1dd5dabad5cc6' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.16.1.200:5060 --->
PUBLISH sip:221@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPjb2d6a25aefc14e9bb5b950a0e9702fb4
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=8a645f6f7d8b44639cc25536e5472683
To: "221" <sip:221@10.16.1.220>
Call-ID: c00c70cb161d4cfab21a5621cf9c57d0
CSeq: 18509 PUBLISH
Event: presence
User-Agent: MicroSIP/3.10.9
Content-Type: application/pidf+xml
Content-Length: 552

<?xml version="1.0" encoding="UTF-8"?>
<presence entity="sip:221@10.16.1.220" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"> <tuple id="pj6186aaa0c8e3495b924db217db734aee"> <status> <basic>open</basic> </status> <timestamp>2016-01-15T16:52:35.491Z</timestamp> <note>Idle</note> </tuple> <dm:person id="pjb43e91373cd54bff99784cb1b05295aa"> <rpid:activities> <rpid:unknown /> </rpid:activities> <dm:note>Idle</dm:note> </dm:person>
</presence>
<------------->
--- (11 headers 3 lines) ---
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPjb2d6a25aefc14e9bb5b950a0e9702fb4;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=8a645f6f7d8b44639cc25536e5472683
To: "221" <sip:221@10.16.1.220>;tag=as2d7bbb91
Call-ID: c00c70cb161d4cfab21a5621cf9c57d0
CSeq: 18509 PUBLISH
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'c00c70cb161d4cfab21a5621cf9c57d0' Method: PUBLISH

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.146:54896 --->


<------------->
Really destroying SIP dialog '1d3a56c2ca1b41d2adc4597bc2faf653' Method: REGISTER

<--- SIP read from UDP:10.16.1.200:5060 --->

<------------->
Really destroying SIP dialog 'e752fc90a1ee4b4aabe1dd5dabad5cc6' Method: REGISTER

<--- SIP read from UDP:10.16.1.146:54896 --->


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

<--- SIP read from UDP:10.16.1.146:54896 --->
INVITE sip:221@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.146:54896;rport;branch=z9hG4bK2137389278
From: <sip:220@10.16.1.220>;tag=556641704
To: <sip:221@10.16.1.220>
Call-ID: 1193192836
CSeq: 20 INVITE
Contact: <sip:220@10.16.1.146:54896>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Max-Forwards: 70
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Subject: Talk
Supported: 100rel, replaces
Content-Length: 197

v=0
o=amsip 382192644 0 IN IP4 10.16.1.146
s=talk
c=IN IP4 10.16.1.146
t=0 0
m=audio 40100 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp-mux
<------------->
--- (14 headers 10 lines) ---
Sending to 10.16.1.146:54896 (no NAT)
Sending to 10.16.1.146:54896 (no NAT)
Using INVITE request as basis request - 1193192836
Found peer '220' for '220' from 10.16.1.146:54896

<--- Reliably Transmitting (no NAT) to 10.16.1.146:54896 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.1.146:54896;branch=z9hG4bK2137389278;received=10.16.1.146;rport=54896
From: <sip:220@10.16.1.220>;tag=556641704
To: <sip:221@10.16.1.220>;tag=as22b98249
Call-ID: 1193192836
CSeq: 20 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2260ef0a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1193192836' in 57920 ms (Method: INVITE)

<--- SIP read from UDP:10.16.1.146:54896 --->
ACK sip:221@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.146:54896;rport;branch=z9hG4bK2137389278
From: <sip:220@10.16.1.220>;tag=556641704
To: <sip:221@10.16.1.220>;tag=as22b98249
Call-ID: 1193192836
CSeq: 20 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.16.1.146:54896 --->
INVITE sip:221@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.146:54896;rport;branch=z9hG4bK1940462086
From: <sip:220@10.16.1.220>;tag=556641704
To: <sip:221@10.16.1.220>
Call-ID: 1193192836
CSeq: 21 INVITE
Contact: <sip:220@10.16.1.146:54896>
Authorization: Digest username="220", realm="asterisk", nonce="2260ef0a", uri="sip:221@10.16.1.220", response="9249fd92698509701ca377e3add11a1b", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Max-Forwards: 70
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Subject: Talk
Supported: 100rel, replaces
Content-Length: 197

v=0
o=amsip 382192644 0 IN IP4 10.16.1.146
s=talk
c=IN IP4 10.16.1.146
t=0 0
m=audio 40100 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp-mux
<------------->
--- (15 headers 10 lines) ---
Sending to 10.16.1.146:54896 (no NAT)
Using INVITE request as basis request - 1193192836
Found peer '220' for '220' from 10.16.1.146:54896
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
failed to extend from 64 to 98
Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), 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|)
Peer audio RTP is at port 10.16.1.146:40100
Looking for 221 in Context_MOBILE (domain 10.16.1.220)
failed to extend from 64 to 98
sip_route_dump: route/path hop: <sip:220@10.16.1.146:54896>

<--- Transmitting (no NAT) to 10.16.1.146:54896 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.1.146:54896;branch=z9hG4bK1940462086;received=10.16.1.146;rport=54896
From: <sip:220@10.16.1.220>;tag=556641704
To: <sip:221@10.16.1.220>
Call-ID: 1193192836
CSeq: 21 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:221@10.16.1.220:5060>
Content-Length: 0


<------------>
failed to extend from 64 to 98
We think we can do text
failed to extend from 64 to 98
Audio is at 12568
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding codec none to SDP
Adding non-codec 0x1 (telephone-event) to SDP
failed to extend from 64 to 98
Reliably Transmitting (no NAT) to 10.16.1.200:5060:
INVITE sip:221@10.16.1.200:5060;ob SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK62157f33
Max-Forwards: 70
From: <sip:220@10.16.1.220>;tag=as28e357b0
To: <sip:221@10.16.1.200:5060;ob>
Contact: <sip:220@10.16.1.220:5060>
Call-ID: 73152fb4099742a51de12ca943175c66@10.16.1.220:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1-cert2
Date: Fri, 15 Jan 2016 15:53:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 900

v=0
o=root 1392895261 1392895261 IN IP4 10.16.1.220
s=Asterisk PBX 13.1-cert2
c=IN IP4 10.16.1.220
t=0 0
m=audio 12568 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

---

<--- SIP read from UDP:10.16.1.200:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.1.220:5060;received=10.16.1.220;branch=z9hG4bK62157f33
Call-ID: 73152fb4099742a51de12ca943175c66@10.16.1.220:5060
From: <sip:220@10.16.1.220>;tag=as28e357b0
To: <sip:221@10.16.1.200;ob>
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:10.16.1.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.220:5060;received=10.16.1.220;branch=z9hG4bK62157f33
Call-ID: 73152fb4099742a51de12ca943175c66@10.16.1.220:5060
From: <sip:220@10.16.1.220>;tag=as28e357b0
To: <sip:221@10.16.1.200;ob>;tag=db99a4ba788545ad80b9c2ee53f7b85a
CSeq: 102 INVITE
Contact: <sip:10.16.1.200:5060>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 3661865592 3661865593 IN IP4 10.16.1.200
s=pjmedia
b=AS:285
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 118 101
c=IN IP4 10.16.1.200
b=TIAS:256000
a=rtcp:4001 IN IP4 10.16.1.200
a=sendrecv
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (11 headers 14 lines) ---
Found RTP audio format 118
Found RTP audio format 101
Found audio description format L16 for ID 118
Found audio description format telephone-event for ID 101
failed to extend from 64 to 98
Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(slin16)/video=(nothing)/text=(nothing), combined - (slin)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.16.1.200:4000
sip_route_dump: route/path hop: <sip:10.16.1.200:5060>
set_destination: Parsing <sip:10.16.1.200:5060> for address/port to send to
set_destination: set destination to 10.16.1.200:5060
Transmitting (no NAT) to 10.16.1.200:5060:
ACK sip:10.16.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK30d1e301
Max-Forwards: 70
From: <sip:220@10.16.1.220>;tag=as28e357b0
To: <sip:221@10.16.1.200:5060;ob>;tag=db99a4ba788545ad80b9c2ee53f7b85a
Contact: <sip:220@10.16.1.220:5060>
Call-ID: 73152fb4099742a51de12ca943175c66@10.16.1.220:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.1-cert2
Content-Length: 0


---
Audio is at 15872
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding codec none to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.16.1.146:54896 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.146:54896;branch=z9hG4bK1940462086;received=10.16.1.146;rport=54896
From: <sip:220@10.16.1.220>;tag=556641704
To: <sip:221@10.16.1.220>;tag=as349c5870
Call-ID: 1193192836
CSeq: 21 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:221@10.16.1.220:5060>
Content-Type: application/sdp
Content-Length: 900

v=0
o=root 1393812786 1393812786 IN IP4 10.16.1.220
s=Asterisk PBX 13.1-cert2
c=IN IP4 10.16.1.220
t=0 0
m=audio 15872 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

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

<--- SIP read from UDP:10.16.1.146:54896 --->
ACK sip:221@10.16.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.146:54896;rport;branch=z9hG4bK1176334093
From: <sip:220@10.16.1.220>;tag=556641704
To: <sip:221@10.16.1.220>;tag=as349c5870
Call-ID: 1193192836
CSeq: 21 ACK
Contact: <sip:220@10.16.1.146:54896>
Max-Forwards: 70
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Reliably Transmitting (no NAT) to 10.16.1.146:54896:
OPTIONS sip:220@10.16.1.146:54896;line=ecbf1a5cf131885 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK0a9ac0c9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.16.1.220>;tag=as681c9971
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>
Contact: <sip:asterisk@10.16.1.220:5060>
Call-ID: 57c43d9075c8266e3d76cf94150c1d5f@10.16.1.220:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1-cert2
Date: Fri, 15 Jan 2016 15:53:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.16.1.146:54896 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK0a9ac0c9
From: "asterisk" <sip:asterisk@10.16.1.220>;tag=as681c9971
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>;tag=1539978412
Call-ID: 57c43d9075c8266e3d76cf94150c1d5f@10.16.1.220:5060
CSeq: 102 OPTIONS
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Accept: application/sdp
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Supported: 100rel, replaces
Content-Length: 383

v=0
o=amsip 0 0 IN IP4 10.16.1.146
s=talk
c=IN IP4 10.16.1.146
t=0 0
m=audio 0 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 118 117 116 115
b=AS:128
a=rtpmap:118 VP8/90000
a=rtpmap:117 H264/90000
a=fmtp:117 profile-level-id=42800c; packetization-mode=1
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000
<------------->
--- (12 headers 16 lines) ---
Really destroying SIP dialog '57c43d9075c8266e3d76cf94150c1d5f@10.16.1.220:5060' Method: OPTIONS

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.146:54896 --->


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

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.200:5060 --->
BYE sip:220@10.16.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPj06ae3b1d2acf47d3a8a5713f37f30d6b
Max-Forwards: 70
From: <sip:221@10.16.1.200;ob>;tag=db99a4ba788545ad80b9c2ee53f7b85a
To: <sip:220@10.16.1.220>;tag=as28e357b0
Call-ID: 73152fb4099742a51de12ca943175c66@10.16.1.220:5060
CSeq: 19912 BYE
User-Agent: MicroSIP/3.10.9
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 10.16.1.200:5060 (no NAT)
Scheduling destruction of SIP dialog '73152fb4099742a51de12ca943175c66@10.16.1.220:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj06ae3b1d2acf47d3a8a5713f37f30d6b;received=10.16.1.200;rport=5060
From: <sip:221@10.16.1.200;ob>;tag=db99a4ba788545ad80b9c2ee53f7b85a
To: <sip:220@10.16.1.220>;tag=as28e357b0
Call-ID: 73152fb4099742a51de12ca943175c66@10.16.1.220:5060
CSeq: 19912 BYE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1193192836' in 57920 ms (Method: ACK)
set_destination: Parsing <sip:220@10.16.1.146:54896> for address/port to send to
set_destination: set destination to 10.16.1.146:54896
Reliably Transmitting (no NAT) to 10.16.1.146:54896:
BYE sip:220@10.16.1.146:54896 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK31ac1844;rport
Max-Forwards: 70
From: <sip:221@10.16.1.220>;tag=as349c5870
To: <sip:220@10.16.1.220>;tag=556641704
Call-ID: 1193192836
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.1-cert2
Proxy-Authorization: Digest username="220", realm="asterisk", algorithm=MD5, uri="sip:10.16.1.220", nonce="2260ef0a", response="53982af76dbf01bb582971d06ad954dc"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:10.16.1.146:54896 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK31ac1844;rport=5060
From: <sip:221@10.16.1.220>;tag=as349c5870
To: <sip:220@10.16.1.220>;tag=556641704
Call-ID: 1193192836
CSeq: 102 BYE
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1193192836' Method: ACK

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.146:54896 --->


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

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.146:54896 --->


<------------->
Really destroying SIP dialog '73152fb4099742a51de12ca943175c66@10.16.1.220:5060' Method: BYE
Reliably Transmitting (no NAT) to 10.16.1.146:54896:
OPTIONS sip:220@10.16.1.146:54896;line=ecbf1a5cf131885 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK6072fd76
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.16.1.220>;tag=as299140d8
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>
Contact: <sip:asterisk@10.16.1.220:5060>
Call-ID: 160ee16e7599df0d31d06c06053e517f@10.16.1.220:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1-cert2
Date: Fri, 15 Jan 2016 15:54:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.16.1.146:54896 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK6072fd76
From: "asterisk" <sip:asterisk@10.16.1.220>;tag=as299140d8
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>;tag=1779743485
Call-ID: 160ee16e7599df0d31d06c06053e517f@10.16.1.220:5060
CSeq: 102 OPTIONS
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Accept: application/sdp
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Supported: 100rel, replaces
Content-Length: 383

v=0
o=amsip 0 0 IN IP4 10.16.1.146
s=talk
c=IN IP4 10.16.1.146
t=0 0
m=audio 0 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 118 117 116 115
b=AS:128
a=rtpmap:118 VP8/90000
a=rtpmap:117 H264/90000
a=fmtp:117 profile-level-id=42800c; packetization-mode=1
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000
<------------->
--- (12 headers 16 lines) ---
Really destroying SIP dialog '160ee16e7599df0d31d06c06053e517f@10.16.1.220:5060' Method: OPTIONS

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.146:54896 --->


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

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.146:54896 --->


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

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.146:54896 --->


<------------->
Reliably Transmitting (no NAT) to 10.16.1.146:54896:
OPTIONS sip:220@10.16.1.146:54896;line=ecbf1a5cf131885 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK4e8e151b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.16.1.220>;tag=as7906085a
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>
Contact: <sip:asterisk@10.16.1.220:5060>
Call-ID: 2213ab3b557cfed47005964723192c46@10.16.1.220:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1-cert2
Date: Fri, 15 Jan 2016 15:55:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.146:54896 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK4e8e151b
From: "asterisk" <sip:asterisk@10.16.1.220>;tag=as7906085a
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>;tag=1247680896
Call-ID: 2213ab3b557cfed47005964723192c46@10.16.1.220:5060
CSeq: 102 OPTIONS
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Accept: application/sdp
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Supported: 100rel, replaces
Content-Length: 383

v=0
o=amsip 0 0 IN IP4 10.16.1.146
s=talk
c=IN IP4 10.16.1.146
t=0 0
m=audio 0 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 118 117 116 115
b=AS:128
a=rtpmap:118 VP8/90000
a=rtpmap:117 H264/90000
a=fmtp:117 profile-level-id=42800c; packetization-mode=1
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000
<------------->
--- (12 headers 16 lines) ---
Really destroying SIP dialog '2213ab3b557cfed47005964723192c46@10.16.1.220:5060' Method: OPTIONS

<--- SIP read from UDP:10.16.1.200:5060 --->
REGISTER sip:10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPj2dafcd355bae409d80dae90ca0dbbf31
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=f5352573f1a5458f8ce3f2885ab5281a
To: "221" <sip:221@10.16.1.220>
Call-ID: e752fc90a1ee4b4aabe1dd5dabad5cc6
CSeq: 40147 REGISTER
User-Agent: MicroSIP/3.10.9
Contact: "221" <sip:221@10.16.1.200:5060;ob>
Expires: 0
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 10.16.1.200:5060 (no NAT)
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj2dafcd355bae409d80dae90ca0dbbf31;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=f5352573f1a5458f8ce3f2885ab5281a
To: "221" <sip:221@10.16.1.220>;tag=as43a845b2
Call-ID: e752fc90a1ee4b4aabe1dd5dabad5cc6
CSeq: 40147 REGISTER
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="739aa166"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e752fc90a1ee4b4aabe1dd5dabad5cc6' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.16.1.200:5060 --->
REGISTER sip:10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPje53690cab2874d99a84d4660d9348d79
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=f5352573f1a5458f8ce3f2885ab5281a
To: "221" <sip:221@10.16.1.220>
Call-ID: e752fc90a1ee4b4aabe1dd5dabad5cc6
CSeq: 40148 REGISTER
User-Agent: MicroSIP/3.10.9
Contact: "221" <sip:221@10.16.1.200:5060;ob>
Expires: 0
Authorization: Digest username="221", realm="asterisk", nonce="739aa166", uri="sip:10.16.1.220", response="19bbfacb59adc8bb6ce4d3613a9ea68c", algorithm=MD5
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPje53690cab2874d99a84d4660d9348d79;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=f5352573f1a5458f8ce3f2885ab5281a
To: "221" <sip:221@10.16.1.220>;tag=as43a845b2
Call-ID: e752fc90a1ee4b4aabe1dd5dabad5cc6
CSeq: 40148 REGISTER
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 0
Date: Fri, 15 Jan 2016 15:55:26 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e752fc90a1ee4b4aabe1dd5dabad5cc6' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.16.1.200:5060 --->
REGISTER sip:10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPj3ad30ae192ca4dfc9002df91f919946f
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=27356535f7f14a728a7a0bda58ffaaf8
To: "221" <sip:221@10.16.1.220>
Call-ID: 1294b8a473c74801848ac85b1dcabcaa
CSeq: 51813 REGISTER
User-Agent: MicroSIP/3.10.9
Contact: "221" <sip:221@10.16.1.200:5060;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 10.16.1.200:5060 (no NAT)
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj3ad30ae192ca4dfc9002df91f919946f;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=27356535f7f14a728a7a0bda58ffaaf8
To: "221" <sip:221@10.16.1.220>;tag=as3ab26814
Call-ID: 1294b8a473c74801848ac85b1dcabcaa
CSeq: 51813 REGISTER
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5cb2a736"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1294b8a473c74801848ac85b1dcabcaa' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.16.1.200:5060 --->
REGISTER sip:10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPj592b1c4dda854f4d84ec3676338834f6
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=27356535f7f14a728a7a0bda58ffaaf8
To: "221" <sip:221@10.16.1.220>
Call-ID: 1294b8a473c74801848ac85b1dcabcaa
CSeq: 51814 REGISTER
User-Agent: MicroSIP/3.10.9
Contact: "221" <sip:221@10.16.1.200:5060;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="221", realm="asterisk", nonce="5cb2a736", uri="sip:10.16.1.220", response="023c160dbebe9fb2d9d3dfc620d52428", algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj592b1c4dda854f4d84ec3676338834f6;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=27356535f7f14a728a7a0bda58ffaaf8
To: "221" <sip:221@10.16.1.220>;tag=as3ab26814
Call-ID: 1294b8a473c74801848ac85b1dcabcaa
CSeq: 51814 REGISTER
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: <sip:221@10.16.1.200:5060;ob>;expires=300
Date: Fri, 15 Jan 2016 15:55:28 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1294b8a473c74801848ac85b1dcabcaa' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.16.1.200:5060 --->
PUBLISH sip:221@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPj3369468b714a40338a036f9a376c036d
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=96d5f3562e564267ba3c31dacdb9c3d3
To: "221" <sip:221@10.16.1.220>
Call-ID: 97c8f9564cae40119564acbed4f7a823
CSeq: 18509 PUBLISH
Event: presence
User-Agent: MicroSIP/3.10.9
Content-Type: application/pidf+xml
Content-Length: 552

<?xml version="1.0" encoding="UTF-8"?>
<presence entity="sip:221@10.16.1.220" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"> <tuple id="pj41c57ea722a34c46ad94e6df8659c8c1"> <status> <basic>open</basic> </status> <timestamp>2016-01-15T16:55:27.235Z</timestamp> <note>Idle</note> </tuple> <dm:person id="pjb6bb1dff5e6742d69d8716af80065df3"> <rpid:activities> <rpid:unknown /> </rpid:activities> <dm:note>Idle</dm:note> </dm:person>
</presence>
<------------->
--- (11 headers 3 lines) ---
Sending to 10.16.1.200:5060 (no NAT)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj3369468b714a40338a036f9a376c036d;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=96d5f3562e564267ba3c31dacdb9c3d3
To: "221" <sip:221@10.16.1.220>;tag=as12124576
Call-ID: 97c8f9564cae40119564acbed4f7a823
CSeq: 18509 PUBLISH
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '97c8f9564cae40119564acbed4f7a823' Method: PUBLISH

<--- SIP read from UDP:10.16.1.146:54896 --->


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

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.146:54896 --->


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

<--- SIP read from UDP:10.16.1.146:54896 --->
INVITE sip:221@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.146:54896;rport;branch=z9hG4bK846368716
From: <sip:220@10.16.1.220>;tag=1348624492
To: <sip:221@10.16.1.220>
Call-ID: 126246262
CSeq: 20 INVITE
Contact: <sip:220@10.16.1.146:54896>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Max-Forwards: 70
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Subject: Talk
Supported: 100rel, replaces
Content-Length: 197

v=0
o=amsip 136212373 0 IN IP4 10.16.1.146
s=talk
c=IN IP4 10.16.1.146
t=0 0
m=audio 40100 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp-mux
<------------->
--- (14 headers 10 lines) ---
Sending to 10.16.1.146:54896 (no NAT)
Sending to 10.16.1.146:54896 (no NAT)
Using INVITE request as basis request - 126246262
Found peer '220' for '220' from 10.16.1.146:54896

<--- Reliably Transmitting (no NAT) to 10.16.1.146:54896 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.1.146:54896;branch=z9hG4bK846368716;received=10.16.1.146;rport=54896
From: <sip:220@10.16.1.220>;tag=1348624492
To: <sip:221@10.16.1.220>;tag=as3598b4dc
Call-ID: 126246262
CSeq: 20 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1babf095"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '126246262' in 50560 ms (Method: INVITE)

<--- SIP read from UDP:10.16.1.146:54896 --->
ACK sip:221@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.146:54896;rport;branch=z9hG4bK846368716
From: <sip:220@10.16.1.220>;tag=1348624492
To: <sip:221@10.16.1.220>;tag=as3598b4dc
Call-ID: 126246262
CSeq: 20 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.16.1.146:54896 --->
INVITE sip:221@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.146:54896;rport;branch=z9hG4bK1160948305
From: <sip:220@10.16.1.220>;tag=1348624492
To: <sip:221@10.16.1.220>
Call-ID: 126246262
CSeq: 21 INVITE
Contact: <sip:220@10.16.1.146:54896>
Authorization: Digest username="220", realm="asterisk", nonce="1babf095", uri="sip:221@10.16.1.220", response="3e4b1efa6f87fa035d9039675a8832ee", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Max-Forwards: 70
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Subject: Talk
Supported: 100rel, replaces
Content-Length: 197

v=0
o=amsip 136212373 0 IN IP4 10.16.1.146
s=talk
c=IN IP4 10.16.1.146
t=0 0
m=audio 40100 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp-mux
<------------->
--- (15 headers 10 lines) ---
Sending to 10.16.1.146:54896 (no NAT)
Using INVITE request as basis request - 126246262
Found peer '220' for '220' from 10.16.1.146:54896
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
failed to extend from 64 to 98
Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), 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|)
Peer audio RTP is at port 10.16.1.146:40100
Looking for 221 in Context_MOBILE (domain 10.16.1.220)
failed to extend from 64 to 98
sip_route_dump: route/path hop: <sip:220@10.16.1.146:54896>

<--- Transmitting (no NAT) to 10.16.1.146:54896 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.1.146:54896;branch=z9hG4bK1160948305;received=10.16.1.146;rport=54896
From: <sip:220@10.16.1.220>;tag=1348624492
To: <sip:221@10.16.1.220>
Call-ID: 126246262
CSeq: 21 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:221@10.16.1.220:5060>
Content-Length: 0


<------------>
failed to extend from 64 to 98
We think we can do text
failed to extend from 64 to 98
Audio is at 14794
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding codec none to SDP
Adding non-codec 0x1 (telephone-event) to SDP
failed to extend from 64 to 98
Reliably Transmitting (no NAT) to 10.16.1.200:5060:
INVITE sip:221@10.16.1.200:5060;ob SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK2fd9c073
Max-Forwards: 70
From: <sip:220@10.16.1.220>;tag=as52865bfd
To: <sip:221@10.16.1.200:5060;ob>
Contact: <sip:220@10.16.1.220:5060>
Call-ID: 61cb19222dcb238f5e2fa00a664b7e63@10.16.1.220:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1-cert2
Date: Fri, 15 Jan 2016 15:55:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 898

v=0
o=root 888579868 888579868 IN IP4 10.16.1.220
s=Asterisk PBX 13.1-cert2
c=IN IP4 10.16.1.220
t=0 0
m=audio 14794 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

---

<--- SIP read from UDP:10.16.1.200:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.1.220:5060;received=10.16.1.220;branch=z9hG4bK2fd9c073
Call-ID: 61cb19222dcb238f5e2fa00a664b7e63@10.16.1.220:5060
From: <sip:220@10.16.1.220>;tag=as52865bfd
To: <sip:221@10.16.1.200;ob>
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:10.16.1.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.220:5060;received=10.16.1.220;branch=z9hG4bK2fd9c073
Call-ID: 61cb19222dcb238f5e2fa00a664b7e63@10.16.1.220:5060
From: <sip:220@10.16.1.220>;tag=as52865bfd
To: <sip:221@10.16.1.200;ob>;tag=736af6300525404da35a9db39842d336
CSeq: 102 INVITE
Contact: <sip:10.16.1.200:5060>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 3661865753 3661865754 IN IP4 10.16.1.200
s=pjmedia
b=AS:285
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 118 101
c=IN IP4 10.16.1.200
b=TIAS:256000
a=rtcp:4001 IN IP4 10.16.1.200
a=sendrecv
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (11 headers 14 lines) ---
Found RTP audio format 118
Found RTP audio format 101
Found audio description format L16 for ID 118
Found audio description format telephone-event for ID 101
failed to extend from 64 to 98
Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(slin16)/video=(nothing)/text=(nothing), combined - (slin)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.16.1.200:4000
sip_route_dump: route/path hop: <sip:10.16.1.200:5060>
set_destination: Parsing <sip:10.16.1.200:5060> for address/port to send to
set_destination: set destination to 10.16.1.200:5060
Transmitting (no NAT) to 10.16.1.200:5060:
ACK sip:10.16.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK1b7395ff
Max-Forwards: 70
From: <sip:220@10.16.1.220>;tag=as52865bfd
To: <sip:221@10.16.1.200:5060;ob>;tag=736af6300525404da35a9db39842d336
Contact: <sip:220@10.16.1.220:5060>
Call-ID: 61cb19222dcb238f5e2fa00a664b7e63@10.16.1.220:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.1-cert2
Content-Length: 0


---
Audio is at 17176
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding codec none to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.16.1.146:54896 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.146:54896;branch=z9hG4bK1160948305;received=10.16.1.146;rport=54896
From: <sip:220@10.16.1.220>;tag=1348624492
To: <sip:221@10.16.1.220>;tag=as3a3f149c
Call-ID: 126246262
CSeq: 21 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:221@10.16.1.220:5060>
Content-Type: application/sdp
Content-Length: 900

v=0
o=root 1378269650 1378269650 IN IP4 10.16.1.220
s=Asterisk PBX 13.1-cert2
c=IN IP4 10.16.1.220
t=0 0
m=audio 17176 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

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

<--- SIP read from UDP:10.16.1.146:54896 --->
ACK sip:221@10.16.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.146:54896;rport;branch=z9hG4bK1903566575
From: <sip:220@10.16.1.220>;tag=1348624492
To: <sip:221@10.16.1.220>;tag=as3a3f149c
Call-ID: 126246262
CSeq: 21 ACK
Contact: <sip:220@10.16.1.146:54896>
Max-Forwards: 70
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Content-Length: 0

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

<--- SIP read from UDP:10.16.1.200:5060 --->

<------------->
Really destroying SIP dialog 'e752fc90a1ee4b4aabe1dd5dabad5cc6' Method: REGISTER
Really destroying SIP dialog '1294b8a473c74801848ac85b1dcabcaa' Method: REGISTER

<--- SIP read from UDP:10.16.1.200:5060 --->
BYE sip:220@10.16.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPjd64144efeb0644b893533030cedde225
Max-Forwards: 70
From: <sip:221@10.16.1.200;ob>;tag=736af6300525404da35a9db39842d336
To: <sip:220@10.16.1.220>;tag=as52865bfd
Call-ID: 61cb19222dcb238f5e2fa00a664b7e63@10.16.1.220:5060
CSeq: 19912 BYE
User-Agent: MicroSIP/3.10.9
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 10.16.1.200:5060 (no NAT)
Scheduling destruction of SIP dialog '61cb19222dcb238f5e2fa00a664b7e63@10.16.1.220:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPjd64144efeb0644b893533030cedde225;received=10.16.1.200;rport=5060
From: <sip:221@10.16.1.200;ob>;tag=736af6300525404da35a9db39842d336
To: <sip:220@10.16.1.220>;tag=as52865bfd
Call-ID: 61cb19222dcb238f5e2fa00a664b7e63@10.16.1.220:5060
CSeq: 19912 BYE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '126246262' in 50560 ms (Method: ACK)
set_destination: Parsing <sip:220@10.16.1.146:54896> for address/port to send to
set_destination: set destination to 10.16.1.146:54896
Reliably Transmitting (no NAT) to 10.16.1.146:54896:
BYE sip:220@10.16.1.146:54896 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK28618890;rport
Max-Forwards: 70
From: <sip:221@10.16.1.220>;tag=as3a3f149c
To: <sip:220@10.16.1.220>;tag=1348624492
Call-ID: 126246262
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.1-cert2
Proxy-Authorization: Digest username="220", realm="asterisk", algorithm=MD5, uri="sip:10.16.1.220", nonce="1babf095", response="9bce38684f21d8d912508290900c9f89"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:10.16.1.146:54896 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK28618890;rport=5060
From: <sip:221@10.16.1.220>;tag=as3a3f149c
To: <sip:220@10.16.1.220>;tag=1348624492
Call-ID: 126246262
CSeq: 102 BYE
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '126246262' Method: ACK

<--- SIP read from UDP:10.16.1.200:5060 --->
INVITE sip:220@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPj8a874f15e77341f290d53d8e7a45c07e
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=48eea5a239a7445687e01d8b37e97238
To: <sip:220@10.16.1.220>
Contact: "221" <sip:221@10.16.1.200:5060;ob>
Call-ID: a8b3d4d35ef84af2879f35fd24ea32b4
CSeq: 28928 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.10.9
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 3661865768 3661865768 IN IP4 10.16.1.200
s=pjmedia
b=AS:285
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 118 101
c=IN IP4 10.16.1.200
b=TIAS:256000
a=rtcp:4001 IN IP4 10.16.1.200
a=sendrecv
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (15 headers 14 lines) ---
Sending to 10.16.1.200:5060 (no NAT)
Sending to 10.16.1.200:5060 (no NAT)
Using INVITE request as basis request - a8b3d4d35ef84af2879f35fd24ea32b4
Found peer '221' for '221' from 10.16.1.200:5060

<--- Reliably Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj8a874f15e77341f290d53d8e7a45c07e;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=48eea5a239a7445687e01d8b37e97238

To: <sip:220@10.16.1.220>;tag=as7a371b87
Call-ID: a8b3d4d35ef84af2879f35fd24ea32b4
CSeq: 28928 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="396916b2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a8b3d4d35ef84af2879f35fd24ea32b4' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:10.16.1.200:5060 --->
ACK sip:220@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPj8a874f15e77341f290d53d8e7a45c07e
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=48eea5a239a7445687e01d8b37e97238
To: <sip:220@10.16.1.220>;tag=as7a371b87
Call-ID: a8b3d4d35ef84af2879f35fd24ea32b4
CSeq: 28928 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.16.1.200:5060 --->
INVITE sip:220@10.16.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPj40085867acbc4f43924ceb88ba4841d7
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=48eea5a239a7445687e01d8b37e97238
To: <sip:220@10.16.1.220>
Contact: "221" <sip:221@10.16.1.200:5060;ob>
Call-ID: a8b3d4d35ef84af2879f35fd24ea32b4
CSeq: 28929 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.10.9
Authorization: Digest username="221", realm="asterisk", nonce="396916b2", uri="sip:220@10.16.1.220", response="307b3b364e8704123ac5f49a5d7c704a", algorithm=MD5
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 3661865768 3661865768 IN IP4 10.16.1.200
s=pjmedia
b=AS:285
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 118 101
c=IN IP4 10.16.1.200
b=TIAS:256000
a=rtcp:4001 IN IP4 10.16.1.200
a=sendrecv
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (16 headers 14 lines) ---
Sending to 10.16.1.200:5060 (no NAT)
Using INVITE request as basis request - a8b3d4d35ef84af2879f35fd24ea32b4
Found peer '221' for '221' from 10.16.1.200:5060
Found RTP audio format 118
Found RTP audio format 101
Found audio description format L16 for ID 118
Found audio description format telephone-event for ID 101
failed to extend from 64 to 98
Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(slin16)/video=(nothing)/text=(nothing), combined - (slin)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.16.1.200:4000
Looking for 220 in Context_MOBILE (domain 10.16.1.220)
failed to extend from 64 to 98
sip_route_dump: route/path hop: <sip:221@10.16.1.200:5060;ob>

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj40085867acbc4f43924ceb88ba4841d7;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=48eea5a239a7445687e01d8b37e97238
To: <sip:220@10.16.1.220>
Call-ID: a8b3d4d35ef84af2879f35fd24ea32b4
CSeq: 28929 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:220@10.16.1.220:5060>
Content-Length: 0


<------------>
failed to extend from 64 to 98
We think we can do text
failed to extend from 64 to 98
Audio is at 11424
Adding codec slin to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding codec none to SDP
Adding non-codec 0x1 (telephone-event) to SDP
failed to extend from 64 to 98
Reliably Transmitting (no NAT) to 10.16.1.146:54896:
INVITE sip:220@10.16.1.146:54896;line=ecbf1a5cf131885 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK7381166f
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=as467c608a
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>
Contact: <sip:221@10.16.1.220:5060>
Call-ID: 62b5f4b04d80377f508000245e954c75@10.16.1.220:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1-cert2
Date: Fri, 15 Jan 2016 15:56:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 900

v=0
o=root 1426782497 1426782497 IN IP4 10.16.1.220
s=Asterisk PBX 13.1-cert2
c=IN IP4 10.16.1.220
t=0 0
m=audio 11424 RTP/AVP 118 0 8 3 4 111 112 5 10 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:118 L16/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

---

<--- SIP read from UDP:10.16.1.146:54896 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK7381166f
From: "221" <sip:221@10.16.1.220>;tag=as467c608a
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>
Call-ID: 62b5f4b04d80377f508000245e954c75@10.16.1.220:5060
CSeq: 102 INVITE
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Content-Length: 0

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

<--- SIP read from UDP:10.16.1.146:54896 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK7381166f
From: "221" <sip:221@10.16.1.220>;tag=as467c608a
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>;tag=2105126077
Call-ID: 62b5f4b04d80377f508000245e954c75@10.16.1.220:5060
CSeq: 102 INVITE
Contact: <sip:220@10.16.1.146:54896>
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Supported: 100rel, replaces
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:220@10.16.1.146:54896>

<--- Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj40085867acbc4f43924ceb88ba4841d7;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=48eea5a239a7445687e01d8b37e97238
To: <sip:220@10.16.1.220>;tag=as758757e3
Call-ID: a8b3d4d35ef84af2879f35fd24ea32b4
CSeq: 28929 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:220@10.16.1.220:5060>
Content-Length: 0


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

<--- SIP read from UDP:10.16.1.146:54896 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK7381166f
From: "221" <sip:221@10.16.1.220>;tag=as467c608a
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>;tag=2105126077
Call-ID: 62b5f4b04d80377f508000245e954c75@10.16.1.220:5060
CSeq: 102 INVITE
Contact: <sip:220@10.16.1.146:54896>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Supported: 100rel, replaces
Content-Length: 198

v=0
o=amsip 1630402631 0 IN IP4 10.16.1.146
s=talk
c=IN IP4 10.16.1.146
t=0 0
m=audio 40100 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=fmtp:101 0-16
<------------->
--- (12 headers 10 lines) ---
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
failed to extend from 64 to 98
Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), 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|)
Peer audio RTP is at port 10.16.1.146:40100
sip_route_dump: route/path hop: <sip:220@10.16.1.146:54896>
set_destination: Parsing <sip:220@10.16.1.146:54896> for address/port to send to
set_destination: set destination to 10.16.1.146:54896
Transmitting (no NAT) to 10.16.1.146:54896:
ACK sip:220@10.16.1.146:54896 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK10ac7513
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=as467c608a
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>;tag=2105126077
Contact: <sip:221@10.16.1.220:5060>
Call-ID: 62b5f4b04d80377f508000245e954c75@10.16.1.220:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.1-cert2
Content-Length: 0


---

<--- SIP read from UDP:10.16.1.146:54896 --->


<------------->
Audio is at 11678
Adding codec slin to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding codec none to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.16.1.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.200:5060;branch=z9hG4bKPj40085867acbc4f43924ceb88ba4841d7;received=10.16.1.200;rport=5060
From: "221" <sip:221@10.16.1.220>;tag=48eea5a239a7445687e01d8b37e97238
To: <sip:220@10.16.1.220>;tag=as758757e3
Call-ID: a8b3d4d35ef84af2879f35fd24ea32b4
CSeq: 28929 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:220@10.16.1.220:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 900

v=0
o=root 1811583576 1811583576 IN IP4 10.16.1.220
s=Asterisk PBX 13.1-cert2
c=IN IP4 10.16.1.220
t=0 0
m=audio 11678 RTP/AVP 118 0 8 3 4 111 112 5 10 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:118 L16/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

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

<--- SIP read from UDP:10.16.1.200:5060 --->
ACK sip:220@10.16.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.200:5060;rport;branch=z9hG4bKPjad76d0f526f84523b95afc934c6ee04c
Max-Forwards: 70
From: "221" <sip:221@10.16.1.220>;tag=48eea5a239a7445687e01d8b37e97238
To: <sip:220@10.16.1.220>;tag=as758757e3
Call-ID: a8b3d4d35ef84af2879f35fd24ea32b4
CSeq: 28929 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.16.1.200:5060 --->

<------------->
Reliably Transmitting (no NAT) to 10.16.1.146:54896:
OPTIONS sip:220@10.16.1.146:54896;line=ecbf1a5cf131885 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK751b025d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.16.1.220>;tag=as4d343929
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>
Contact: <sip:asterisk@10.16.1.220:5060>
Call-ID: 799d4031405c12c50e54b97c6395bc26@10.16.1.220:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1-cert2
Date: Fri, 15 Jan 2016 15:56:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.16.1.146:54896 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK751b025d
From: "asterisk" <sip:asterisk@10.16.1.220>;tag=as4d343929
To: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>;tag=1933739672
Call-ID: 799d4031405c12c50e54b97c6395bc26@10.16.1.220:5060
CSeq: 102 OPTIONS
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Accept: application/sdp
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Supported: 100rel, replaces
Content-Length: 383

v=0
o=amsip 0 0 IN IP4 10.16.1.146
s=talk
c=IN IP4 10.16.1.146
t=0 0
m=audio 0 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 118 117 116 115
b=AS:128
a=rtpmap:118 VP8/90000
a=rtpmap:117 H264/90000
a=fmtp:117 profile-level-id=42800c; packetization-mode=1
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000
<------------->
--- (12 headers 16 lines) ---
Really destroying SIP dialog '799d4031405c12c50e54b97c6395bc26@10.16.1.220:5060' Method: OPTIONS

<--- SIP read from UDP:10.16.1.146:54896 --->
BYE sip:221@10.16.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.1.146:54896;rport;branch=z9hG4bK1600077565
From: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>;tag=2105126077
To: "221" <sip:221@10.16.1.220>;tag=as467c608a
Call-ID: 62b5f4b04d80377f508000245e954c75@10.16.1.220:5060
CSeq: 2 BYE
Max-Forwards: 70
User-Agent: antisip/5.1.0-581-gc343433-Dec-17-2015 amdroid/4.2.9 JY-G3/4.2.1
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 10.16.1.146:54896 (no NAT)
Scheduling destruction of SIP dialog '62b5f4b04d80377f508000245e954c75@10.16.1.220:5060' in 50560 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.16.1.146:54896 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.146:54896;branch=z9hG4bK1600077565;received=10.16.1.146;rport=54896
From: <sip:220@10.16.1.146:54896;line=ecbf1a5cf131885>;tag=2105126077
To: "221" <sip:221@10.16.1.220>;tag=as467c608a
Call-ID: 62b5f4b04d80377f508000245e954c75@10.16.1.220:5060
CSeq: 2 BYE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a8b3d4d35ef84af2879f35fd24ea32b4' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:221@10.16.1.200:5060;ob> for address/port to send to
set_destination: set destination to 10.16.1.200:5060
Reliably Transmitting (no NAT) to 10.16.1.200:5060:
BYE sip:221@10.16.1.200:5060;ob SIP/2.0
Via: SIP/2.0/UDP 10.16.1.220:5060;branch=z9hG4bK28621cfc;rport
Max-Forwards: 70
From: <sip:220@10.16.1.220>;tag=as758757e3
To: "221" <sip:221@10.16.1.220>;tag=48eea5a239a7445687e01d8b37e97238
Call-ID: a8b3d4d35ef84af2879f35fd24ea32b4
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.1-cert2
Proxy-Authorization: Digest username="221", realm="asterisk", algorithm=MD5, uri="sip:10.16.1.220", nonce="396916b2", response="b7c9c38964329ca28ec1fd8477d2064c"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:10.16.1.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.1.220:5060;rport=5060;received=10.16.1.220;branch=z9hG4bK28621cfc
Call-ID: a8b3d4d35ef84af2879f35fd24ea32b4
From: <sip:220@10.16.1.220>;tag=as758757e3
To: "221" <sip:221@10.16.1.220>;tag=48eea5a239a7445687e01d8b37e97238
CSeq: 102 BYE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'a8b3d4d35ef84af2879f35fd24ea32b4' Method: ACK

<--- SIP read from UDP:10.16.1.200:5060 --->

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

<--- SIP read from UDP:10.16.1.146:54896 --->


<------------->
Really destroying SIP dialog '61cb19222dcb238f5e2fa00a664b7e63@10.16.1.220:5060' Method: BYE
No such command '' (type 'core show help ' for other possible commands)

<--- SIP read from UDP:10.16.1.200:5060 --->

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