Last login: Tue Aug 23 19:20:14 on ttys000
jeremiahs-MacBook-Pro:~ jeremiahbooker$ ssh root@110.5.42.156
root@110.5.42.156's password:
Last login: Tue Aug 23 19:30:38 2016 from 183.76.169.117
 _____              ____  ______  __
|  ___| __ ___  ___|  _ \| __ ) \/ /
| |_ | '__/ _ \/ _ \ |_) |  _ \\  /
|  _|| | |  __/  __/  __/| |_) /  \
|_|  |_|  \___|\___|_|   |____/_/\_\

NOTICE! You have 4 notifications! Please log into the UI to see them!

Current Network Configuration
+-----------+-------------------+--------------------------------------+
| Interface | MAC Address       | IP Addresses                         |
+-----------+-------------------+--------------------------------------+
| eth0      | 00:50:43:01:72:99 | 192.168.1.3                          |
|           |                   | 2408:210:725:f900:250:43ff:fe01:7299 |
|           |                   | fe80::250:43ff:fe01:7299             |
+-----------+-------------------+--------------------------------------+

Please note most tasks should be handled through the GUI.
You can access the GUI by typing one of the above IPs in to your web browser.
For support please visit:
    http://www.freepbx.org/support-and-professional-services

[root@localhost ~]# asterisk -rcvvvvv
Asterisk 11.23.0, Copyright (C) 1999 - 2013 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 11.23.0 currently running on localhost (pid = 13421)

<--- SIP read from UDP:37.187.164.39:5070 --->
INVITE sip:001972597751296@110.5.42.156 SIP/2.0
To: 001972597751296<sip:001972597751296@110.5.42.156>
From: 2222<sip:2222@110.5.42.156>;tag=aa344755
Via: SIP/2.0/UDP 37.187.164.39:5070;branch=z9hG4bK-c264a7e6e5c9d5e88d5220c26a94d1d3;rport
Call-ID: c264a7e6e5c9d5e88d5220c26a94d1d3
CSeq: 1 INVITE
Contact: <sip:2222@37.187.164.39:5070>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE
User-Agent: sipcli/v1.8
Content-Type: application/sdp
Content-Length: 281

v=0
o=sipcli-Session 28577513 1970755320 IN IP4 37.187.164.39
s=sipcli
c=IN IP4 37.187.164.39
t=0 0
m=audio 5072 RTP/AVP 18 0 8 101
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 13 lines) ---
Sending to 37.187.164.39:5070 (NAT)
Sending to 37.187.164.39:5070 (NAT)
Using INVITE request as basis request - c264a7e6e5c9d5e88d5220c26a94d1d3
No matching peer for '2222' from '37.187.164.39:5070'
  == Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 37.187.164.39:5072
Looking for 001972597751296 in from-sip-external (domain 110.5.42.156)
list_route: hop: <sip:2222@37.187.164.39:5070>

<--- Transmitting (NAT) to 37.187.164.39:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 37.187.164.39:5070;branch=z9hG4bK-c264a7e6e5c9d5e88d5220c26a94d1d3;received=37.187.164.39;rport=5070
From: 2222<sip:2222@110.5.42.156>;tag=aa344755
To: 001972597751296<sip:001972597751296@110.5.42.156>
Call-ID: c264a7e6e5c9d5e88d5220c26a94d1d3
CSeq: 1 INVITE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:001972597751296@110.5.42.156:5060>
Content-Length: 0


<------------>
    -- Executing [001972597751296@from-sip-external:1] NoOp("SIP/110.5.42.156-000002b0", "Received incoming SIP connection from unknown peer to 001972597751296") in new stack
    -- Executing [001972597751296@from-sip-external:2] Set("SIP/110.5.42.156-000002b0", "DID=001972597751296") in new stack
    -- Executing [001972597751296@from-sip-external:3] Goto("SIP/110.5.42.156-000002b0", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/110.5.42.156-000002b0", "1?checklang:noanonymous") in new stack
    -- Goto (from-sip-external,s,2)
    -- Executing [s@from-sip-external:2] GotoIf("SIP/110.5.42.156-000002b0", "0?setlanguage:from-trunk,001972597751296,1") in new stack
    -- Goto (from-trunk,001972597751296,1)
    -- Executing [001972597751296@from-trunk:1] Set("SIP/110.5.42.156-000002b0", "__FROM_DID=001972597751296") in new stack
    -- Executing [001972597751296@from-trunk:2] NoOp("SIP/110.5.42.156-000002b0", "Received an unknown call with DID set to 001972597751296") in new stack
    -- Executing [001972597751296@from-trunk:3] Goto("SIP/110.5.42.156-000002b0", "s,a2") in new stack
    -- Goto (from-trunk,s,2)
    -- Executing [s@from-trunk:2] Answer("SIP/110.5.42.156-000002b0", "") in new stack
Audio is at 18892
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 37.187.164.39:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 37.187.164.39:5070;branch=z9hG4bK-c264a7e6e5c9d5e88d5220c26a94d1d3;received=37.187.164.39;rport=5070
From: 2222<sip:2222@110.5.42.156>;tag=aa344755
To: 001972597751296<sip:001972597751296@110.5.42.156>;tag=as6a609d37
Call-ID: c264a7e6e5c9d5e88d5220c26a94d1d3
CSeq: 1 INVITE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:001972597751296@110.5.42.156:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1810499285 1810499285 IN IP4 110.5.42.156
s=Asterisk PBX 11.23.0
c=IN IP4 110.5.42.156
t=0 0
m=audio 18892 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:37.187.164.39:5070 --->
ACK sip:001972597751296@110.5.42.156:5060 SIP/2.0
Via: SIP/2.0/UDP 37.187.164.39:5070;branch=z9hG4bK-c264a7e6e5c9d5e88d5220c26a94d1d3;rport
From: 2222<sip:2222@110.5.42.156>;tag=aa344755
To: 001972597751296<sip:001972597751296@110.5.42.156>;tag=as6a609d37
Call-ID: c264a7e6e5c9d5e88d5220c26a94d1d3
CSeq: 1 ACK
Contact: <sip:2222@37.187.164.39:5070>
Max-Forwards: 70
Content-Length: 0

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

<--- SIP read from UDP:37.187.164.39:5070 --->
BYE sip:001972597751296@110.5.42.156:5060 SIP/2.0
Via: SIP/2.0/UDP 37.187.164.39:5070;branch=z9hG4bK-c264a7e6e5c9d5e88d5220c26a94d1d3;rport
From: 2222<sip:2222@110.5.42.156>;tag=aa344755
To: 001972597751296<sip:001972597751296@110.5.42.156>;tag=as6a609d37
Call-ID: c264a7e6e5c9d5e88d5220c26a94d1d3
CSeq: 2 BYE
Contact: <sip:2222@37.187.164.39:5070>
Max-Forwards: 70
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 37.187.164.39:5070 (NAT)
Scheduling destruction of SIP dialog 'c264a7e6e5c9d5e88d5220c26a94d1d3' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 37.187.164.39:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 37.187.164.39:5070;branch=z9hG4bK-c264a7e6e5c9d5e88d5220c26a94d1d3;received=37.187.164.39;rport=5070
From: 2222<sip:2222@110.5.42.156>;tag=aa344755
To: 001972597751296<sip:001972597751296@110.5.42.156>;tag=as6a609d37
Call-ID: c264a7e6e5c9d5e88d5220c26a94d1d3
CSeq: 2 BYE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (from-trunk, s, 2) exited non-zero on 'SIP/110.5.42.156-000002b0'
    -- Executing [h@from-trunk:1] Macro("SIP/110.5.42.156-000002b0", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] ExecIf("SIP/110.5.42.156-000002b0", "0?Set(CDR(recordingfile)=.)") in new stack
    -- Executing [s@macro-hangupcall:2] GotoIf("SIP/110.5.42.156-000002b0", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] ExecIf("SIP/110.5.42.156-000002b0", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:5] Hangup("SIP/110.5.42.156-000002b0", "") in new stack
  == Spawn extension (macro-hangupcall, s, 5) exited non-zero on 'SIP/110.5.42.156-000002b0' in macro 'hangupcall'
  == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/110.5.42.156-000002b0'
localhost*CLI> sip set debug on
SIP Debugging re-enabled
Reliably Transmitting (NAT) to 183.76.169.117:40408:
OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK36298e26;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as70eaeaf0
To: <sip:200@10.0.1.34:5060>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 11b9754108afb2834d20fd150bb6f7fc@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:33:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:183.76.169.117:40408 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK36298e26;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as70eaeaf0
To: <sip:200@10.0.1.34:5060>;tag=130327992
Call-ID: 11b9754108afb2834d20fd150bb6f7fc@110.5.42.156:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.2.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '11b9754108afb2834d20fd150bb6f7fc@110.5.42.156:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 172.16.71.188:5060:
OPTIONS sip:172.16.71.188 SIP/2.0
Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK283b13f6;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as496a9c9b
To: <sip:172.16.71.188>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 5f24b90b2da44df9178e14cf21510aa7@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:33:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:172.16.71.188:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK283b13f6;received=192.168.1.3;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as496a9c9b
To: <sip:172.16.71.188>;tag=as4d485d9f
Call-ID: 5f24b90b2da44df9178e14cf21510aa7@110.5.42.156:5060
CSeq: 102 OPTIONS
Server: MCAT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '5f24b90b2da44df9178e14cf21510aa7@110.5.42.156:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 172.16.71.189:5060:
OPTIONS sip:172.16.71.189 SIP/2.0
Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK04275c41;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as1f7de0e8
To: <sip:172.16.71.189>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 521185eb4dd94eee4235dc7167ccbb82@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:33:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:172.16.71.189:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK04275c41;received=192.168.1.3;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as1f7de0e8
To: <sip:172.16.71.189>;tag=as6fa0a0c0
Call-ID: 521185eb4dd94eee4235dc7167ccbb82@110.5.42.156:5060
CSeq: 102 OPTIONS
Server: MCAT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '521185eb4dd94eee4235dc7167ccbb82@110.5.42.156:5060' Method: OPTIONS

<--- SIP read from UDP:172.16.71.189:5060 --->
OPTIONS sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.71.189:5060;branch=z9hG4bK57a1f7dd
Max-Forwards: 70
From: "MCAT" <sip:MCAT@172.16.71.189>;tag=as6f0e92e5
To: <sip:192.168.1.3>
Contact: <sip:MCAT@172.16.71.189:5060>
Call-ID: 72e30ca54de4573f4920c74f0dcb778e@172.16.71.189:5060
CSeq: 102 OPTIONS
User-Agent: MCAT
Date: Tue, 23 Aug 2016 10:33:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 172.16.71.189:5060 (NAT)
Looking for s in from-sip-external (domain 192.168.1.3)

<--- Transmitting (NAT) to 172.16.71.189:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.71.189:5060;branch=z9hG4bK57a1f7dd;received=172.16.71.189;rport=5060
From: "MCAT" <sip:MCAT@172.16.71.189>;tag=as6f0e92e5
To: <sip:192.168.1.3>;tag=as12bb127b
Call-ID: 72e30ca54de4573f4920c74f0dcb778e@172.16.71.189:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:110.5.42.156:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '72e30ca54de4573f4920c74f0dcb778e@172.16.71.189:5060' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:183.76.169.117:40408 --->
INVITE sip:09016192354@110.5.42.156 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK911597809;rport
From: <sip:200@110.5.42.156>;tag=871381661
To: <sip:09016192354@110.5.42.156>
Call-ID: 268191048-5060-18@BA.A.B.DE
CSeq: 170 INVITE
Contact: <sip:200@10.0.1.34:5060>
Max-Forwards: 70
User-Agent: Grandstream GXP1625 1.0.2.27
Privacy: none
P-Preferred-Identity: <sip:200@110.5.42.156>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 326

v=0
o=200 8000 8000 IN IP4 10.0.1.34
s=SIP Call
c=IN IP4 10.0.1.34
t=0 0
m=audio 5004 RTP/AVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (16 headers 16 lines) ---
Sending to 183.76.169.117:40408 (NAT)
Sending to 183.76.169.117:40408 (NAT)
Using INVITE request as basis request - 268191048-5060-18@BA.A.B.DE
Found peer '200' for '200' from 183.76.169.117:40408

<--- Reliably Transmitting (NAT) to 183.76.169.117:40408 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK911597809;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=871381661
To: <sip:09016192354@110.5.42.156>;tag=as6a1ca082
Call-ID: 268191048-5060-18@BA.A.B.DE
CSeq: 170 INVITE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63e30632"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '268191048-5060-18@BA.A.B.DE' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:183.76.169.117:40408 --->
ACK sip:09016192354@110.5.42.156 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK911597809;rport
From: <sip:200@110.5.42.156>;tag=871381661
To: <sip:09016192354@110.5.42.156>;tag=as6a1ca082
Call-ID: 268191048-5060-18@BA.A.B.DE
CSeq: 170 ACK
Content-Length: 0

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

<--- SIP read from UDP:183.76.169.117:40408 --->
INVITE sip:09016192354@110.5.42.156 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2099182397;rport
From: <sip:200@110.5.42.156>;tag=871381661
To: <sip:09016192354@110.5.42.156>
Call-ID: 268191048-5060-18@BA.A.B.DE
CSeq: 171 INVITE
Contact: <sip:200@10.0.1.34:5060>
Authorization: Digest username="200", realm="asterisk", nonce="63e30632", uri="sip:09016192354@110.5.42.156", response="cdedfcdb3e4a84b600a43f4eefb45e65", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXP1625 1.0.2.27
Privacy: none
P-Preferred-Identity: <sip:200@110.5.42.156>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 326

v=0
o=200 8000 8000 IN IP4 10.0.1.34
s=SIP Call
c=IN IP4 10.0.1.34
t=0 0
m=audio 5004 RTP/AVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 16 lines) ---
Sending to 183.76.169.117:40408 (NAT)
Using INVITE request as basis request - 268191048-5060-18@BA.A.B.DE
Found peer '200' for '200' from 183.76.169.117:40408
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.1.34:5004
Looking for 09016192354 in from-internal (domain 110.5.42.156)
list_route: hop: <sip:200@10.0.1.34:5060>

<--- Transmitting (NAT) to 183.76.169.117:40408 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2099182397;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=871381661
To: <sip:09016192354@110.5.42.156>
Call-ID: 268191048-5060-18@BA.A.B.DE
CSeq: 171 INVITE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:09016192354@110.5.42.156:5060>
Content-Length: 0


<------------>
    -- Executing [09016192354@from-internal:1] ResetCDR("SIP/200-000002b1", "") in new stack
    -- Executing [09016192354@from-internal:2] NoCDR("SIP/200-000002b1", "") in new stack
    -- Executing [09016192354@from-internal:3] Progress("SIP/200-000002b1", "") in new stack
Audio is at 16840
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 183.76.169.117:40408 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2099182397;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=871381661
To: <sip:09016192354@110.5.42.156>;tag=as47194924
Call-ID: 268191048-5060-18@BA.A.B.DE
CSeq: 171 INVITE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:09016192354@110.5.42.156:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 260

v=0
o=root 1847203022 1847203022 IN IP4 110.5.42.156
s=Asterisk PBX 11.23.0
c=IN IP4 110.5.42.156
t=0 0
m=audio 16840 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
    -- Executing [09016192354@from-internal:4] Wait("SIP/200-000002b1", "1") in new stack
       > 0x7f2edc022190 -- Probation passed - setting RTP source address to 183.76.169.117:41029
       > 0x7f2edc022190 -- Probation passed - setting RTP source address to 183.76.169.117:41029
    -- Executing [09016192354@from-internal:5] Playback("SIP/200-000002b1", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    -- <SIP/200-000002b1> Playing 'silence/1.alaw' (language 'en')
    -- <SIP/200-000002b1> Playing 'cannot-complete-as-dialed.alaw' (language 'en')

<--- SIP read from UDP:172.16.71.188:5060 --->
OPTIONS sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.71.188:5060;branch=z9hG4bK4cbc2081
Max-Forwards: 70
From: "MCAT" <sip:MCAT@172.16.71.188>;tag=as6e5f6954
To: <sip:192.168.1.3>
Contact: <sip:MCAT@172.16.71.188:5060>
Call-ID: 0a132e2b2317c6b248ac86b247b17754@172.16.71.188:5060
CSeq: 102 OPTIONS
User-Agent: MCAT
Date: Tue, 23 Aug 2016 10:33:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 172.16.71.188:5060 (NAT)
Looking for s in from-sip-external (domain 192.168.1.3)

<--- Transmitting (NAT) to 172.16.71.188:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.71.188:5060;branch=z9hG4bK4cbc2081;received=172.16.71.188;rport=5060
From: "MCAT" <sip:MCAT@172.16.71.188>;tag=as6e5f6954
To: <sip:192.168.1.3>;tag=as19ed9abf
Call-ID: 0a132e2b2317c6b248ac86b247b17754@172.16.71.188:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:110.5.42.156:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0a132e2b2317c6b248ac86b247b17754@172.16.71.188:5060' in 32000 ms (Method: OPTIONS)
    -- <SIP/200-000002b1> Playing 'check-number-dial-again.alaw' (language 'en')

<--- SIP read from UDP:183.76.169.117:40408 --->
CANCEL sip:09016192354@110.5.42.156 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2099182397;rport
From: <sip:200@110.5.42.156>;tag=871381661
To: <sip:09016192354@110.5.42.156>
Call-ID: 268191048-5060-18@BA.A.B.DE
CSeq: 171 CANCEL
Max-Forwards: 70
User-Agent: Grandstream GXP1625 1.0.2.27
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 183.76.169.117:40408 (NAT)

<--- Reliably Transmitting (NAT) to 183.76.169.117:40408 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2099182397;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=871381661
To: <sip:09016192354@110.5.42.156>;tag=as47194924
Call-ID: 268191048-5060-18@BA.A.B.DE
CSeq: 171 INVITE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (NAT) to 183.76.169.117:40408 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2099182397;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=871381661
To: <sip:09016192354@110.5.42.156>;tag=as47194924
Call-ID: 268191048-5060-18@BA.A.B.DE
CSeq: 171 CANCEL
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (from-internal, 09016192354, 5) exited non-zero on 'SIP/200-000002b1'
    -- Executing [h@from-internal:1] Macro("SIP/200-000002b1", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ExecIf("SIP/200-000002b1", "0?Set(CDR(recordingfile)=.)") in new stack
    -- Executing [s@macro-hangupcall:2] GotoIf("SIP/200-000002b1", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] ExecIf("SIP/200-000002b1", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:5] Hangup("SIP/200-000002b1", "") in new stack
  == Spawn extension (macro-hangupcall, s, 5) exited non-zero on 'SIP/200-000002b1' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-000002b1'

<--- SIP read from UDP:183.76.169.117:40408 --->
ACK sip:09016192354@110.5.42.156 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2099182397;rport
From: <sip:200@110.5.42.156>;tag=871381661
To: <sip:09016192354@110.5.42.156>;tag=as47194924
Call-ID: 268191048-5060-18@BA.A.B.DE
CSeq: 171 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '268191048-5060-18@BA.A.B.DE' Method: ACK
Reliably Transmitting (NAT) to 183.76.169.117:40408: