root@110.5.42.156's password:
Last login: Tue Aug 23 19:39:23 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 -vrrr
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)
localhost*CLI> sip set debug ip 183.76.169.117
SIP Debugging Enabled for IP: 183.76.169.117
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=z9hG4bK731c3d4e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as604657e0
To: <sip:200@10.0.1.34:5060>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 262519634834796b58439f5219748bb1@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:53:22 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=z9hG4bK731c3d4e;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as604657e0
To: <sip:200@10.0.1.34:5060>;tag=118288961
Call-ID: 262519634834796b58439f5219748bb1@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 '262519634834796b58439f5219748bb1@110.5.42.156:5060' 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=z9hG4bK1432922524;rport
From: <sip:200@110.5.42.156>;tag=124288931
To: <sip:09016192354@110.5.42.156>
Call-ID: 353377531-5060-19@BA.A.B.DE
CSeq: 180 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 - 353377531-5060-19@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=z9hG4bK1432922524;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=124288931
To: <sip:09016192354@110.5.42.156>;tag=as0ec8c193
Call-ID: 353377531-5060-19@BA.A.B.DE
CSeq: 180 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="1adb0743"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '353377531-5060-19@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=z9hG4bK1432922524;rport
From: <sip:200@110.5.42.156>;tag=124288931
To: <sip:09016192354@110.5.42.156>;tag=as0ec8c193
Call-ID: 353377531-5060-19@BA.A.B.DE
CSeq: 180 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=z9hG4bK1646421649;rport
From: <sip:200@110.5.42.156>;tag=124288931
To: <sip:09016192354@110.5.42.156>
Call-ID: 353377531-5060-19@BA.A.B.DE
CSeq: 181 INVITE
Contact: <sip:200@10.0.1.34:5060>
Authorization: Digest username="200", realm="asterisk", nonce="1adb0743", uri="sip:09016192354@110.5.42.156", response="69b2cd8c9dddb7ad4f325d52f6a83935", 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 - 353377531-5060-19@BA.A.B.DE
Found peer '200' for '200' from 183.76.169.117:40408
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=z9hG4bK1646421649;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=124288931
To: <sip:09016192354@110.5.42.156>
Call-ID: 353377531-5060-19@BA.A.B.DE
CSeq: 181 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


<------------>
Audio is at 13462
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=z9hG4bK1646421649;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=124288931
To: <sip:09016192354@110.5.42.156>;tag=as5b831fe6
Call-ID: 353377531-5060-19@BA.A.B.DE
CSeq: 181 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 2024175327 2024175327 IN IP4 110.5.42.156
s=Asterisk PBX 11.23.0
c=IN IP4 110.5.42.156
t=0 0
m=audio 13462 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

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

<--- 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=z9hG4bK1646421649;rport
From: <sip:200@110.5.42.156>;tag=124288931
To: <sip:09016192354@110.5.42.156>
Call-ID: 353377531-5060-19@BA.A.B.DE
CSeq: 181 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=z9hG4bK1646421649;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=124288931
To: <sip:09016192354@110.5.42.156>;tag=as5b831fe6
Call-ID: 353377531-5060-19@BA.A.B.DE
CSeq: 181 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=z9hG4bK1646421649;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=124288931
To: <sip:09016192354@110.5.42.156>;tag=as5b831fe6
Call-ID: 353377531-5060-19@BA.A.B.DE
CSeq: 181 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


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

<--- 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=z9hG4bK1646421649;rport
From: <sip:200@110.5.42.156>;tag=124288931
To: <sip:09016192354@110.5.42.156>;tag=as5b831fe6
Call-ID: 353377531-5060-19@BA.A.B.DE
CSeq: 181 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '353377531-5060-19@BA.A.B.DE' Method: ACK
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=z9hG4bK79a8f1f7;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as35853acc
To: <sip:200@10.0.1.34:5060>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 7ceca587177a4b667baafb0f6b6d7bc0@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:53:42 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=z9hG4bK79a8f1f7;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as35853acc
To: <sip:200@10.0.1.34:5060>;tag=1732081387
Call-ID: 7ceca587177a4b667baafb0f6b6d7bc0@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 '7ceca587177a4b667baafb0f6b6d7bc0@110.5.42.156:5060' Method: OPTIONS
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=z9hG4bK3dbf42c9;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as15defe77
To: <sip:200@10.0.1.34:5060>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 177d0b550fd8667e052c7fb120927346@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:54:02 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=z9hG4bK3dbf42c9;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as15defe77
To: <sip:200@10.0.1.34:5060>;tag=299104229
Call-ID: 177d0b550fd8667e052c7fb120927346@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 '177d0b550fd8667e052c7fb120927346@110.5.42.156:5060' Method: OPTIONS
localhost*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@localhost ~]# clear all
[root@localhost ~]# asterisk -vrrr
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)
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=z9hG4bK4908b718;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as3ec5d9fe
To: <sip:200@10.0.1.34:5060>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 510bd5c2477b36350ec879d06c757424@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:54:22 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=z9hG4bK4908b718;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as3ec5d9fe
To: <sip:200@10.0.1.34:5060>;tag=1739782339
Call-ID: 510bd5c2477b36350ec879d06c757424@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 '510bd5c2477b36350ec879d06c757424@110.5.42.156:5060' Method: OPTIONS
localhost*CLI> sip set debug ip 183.76.169.117
SIP Debugging Enabled for IP: 183.76.169.117

<--- 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=z9hG4bK589886018;rport
From: <sip:200@110.5.42.156>;tag=1325728154
To: <sip:09016192354@110.5.42.156>
Call-ID: 129941210-5060-20@BA.A.B.DE
CSeq: 190 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 - 129941210-5060-20@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=z9hG4bK589886018;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=1325728154
To: <sip:09016192354@110.5.42.156>;tag=as14fb3e71
Call-ID: 129941210-5060-20@BA.A.B.DE
CSeq: 190 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="35fdb1f7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '129941210-5060-20@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=z9hG4bK589886018;rport
From: <sip:200@110.5.42.156>;tag=1325728154
To: <sip:09016192354@110.5.42.156>;tag=as14fb3e71
Call-ID: 129941210-5060-20@BA.A.B.DE
CSeq: 190 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=z9hG4bK990693629;rport
From: <sip:200@110.5.42.156>;tag=1325728154
To: <sip:09016192354@110.5.42.156>
Call-ID: 129941210-5060-20@BA.A.B.DE
CSeq: 191 INVITE
Contact: <sip:200@10.0.1.34:5060>
Authorization: Digest username="200", realm="asterisk", nonce="35fdb1f7", uri="sip:09016192354@110.5.42.156", response="819fcc1b28b0f0d6baa5127be7a79dff", 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 - 129941210-5060-20@BA.A.B.DE
Found peer '200' for '200' from 183.76.169.117:40408
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=z9hG4bK990693629;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=1325728154
To: <sip:09016192354@110.5.42.156>
Call-ID: 129941210-5060-20@BA.A.B.DE
CSeq: 191 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


<------------>
Audio is at 12912
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=z9hG4bK990693629;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=1325728154
To: <sip:09016192354@110.5.42.156>;tag=as25f70ace
Call-ID: 129941210-5060-20@BA.A.B.DE
CSeq: 191 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: 256

v=0
o=root 48464541 48464541 IN IP4 110.5.42.156
s=Asterisk PBX 11.23.0
c=IN IP4 110.5.42.156
t=0 0
m=audio 12912 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

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

<--- Reliably Transmitting (NAT) to 183.76.169.117:40408 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK990693629;received=183.76.169.117;rport=40408
From: <sip:200@110.5.42.156>;tag=1325728154
To: <sip:09016192354@110.5.42.156>;tag=as25f70ace
Call-ID: 129941210-5060-20@BA.A.B.DE
CSeq: 191 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
Content-Length: 0


<------------>
[2016-08-23 19:54:40] WARNING[6331][C-000002c4]: channel.c:4861 ast_prod: Prodding channel 'SIP/200-000002b5' failed

<--- 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=z9hG4bK990693629;rport
From: <sip:200@110.5.42.156>;tag=1325728154
To: <sip:09016192354@110.5.42.156>;tag=as25f70ace
Call-ID: 129941210-5060-20@BA.A.B.DE
CSeq: 191 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '129941210-5060-20@BA.A.B.DE' Method: ACK
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=z9hG4bK2a3a2957;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as4285ba66
To: <sip:200@10.0.1.34:5060>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 152671a26408a8302fe9092928fa709f@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:54:42 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=z9hG4bK2a3a2957;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as4285ba66
To: <sip:200@10.0.1.34:5060>;tag=336763972
Call-ID: 152671a26408a8302fe9092928fa709f@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 '152671a26408a8302fe9092928fa709f@110.5.42.156:5060' Method: OPTIONS
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=z9hG4bK698c4317;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as395026cc
To: <sip:200@10.0.1.34:5060>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 32358b4c33a4fe2411320a71765318a8@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:55:02 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=z9hG4bK698c4317;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as395026cc
To: <sip:200@10.0.1.34:5060>;tag=1223370465
Call-ID: 32358b4c33a4fe2411320a71765318a8@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 '32358b4c33a4fe2411320a71765318a8@110.5.42.156:5060' Method: OPTIONS
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=z9hG4bK5b6ecb6c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as5a085fad
To: <sip:200@10.0.1.34:5060>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 4eb7e07e6707f10b6ec86ef51d8294bf@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:55:22 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=z9hG4bK5b6ecb6c;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as5a085fad
To: <sip:200@10.0.1.34:5060>;tag=634799037
Call-ID: 4eb7e07e6707f10b6ec86ef51d8294bf@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 '4eb7e07e6707f10b6ec86ef51d8294bf@110.5.42.156:5060' Method: OPTIONS
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=z9hG4bK4c45e42b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as208e5ab7
To: <sip:200@10.0.1.34:5060>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 39d34fb37b8346e35909d83b6d6535bb@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:55:42 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=z9hG4bK4c45e42b;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as208e5ab7
To: <sip:200@10.0.1.34:5060>;tag=314099702
Call-ID: 39d34fb37b8346e35909d83b6d6535bb@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 '39d34fb37b8346e35909d83b6d6535bb@110.5.42.156:5060' Method: OPTIONS
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=z9hG4bK118080c0;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as6d1b2a20
To: <sip:200@10.0.1.34:5060>
Contact: <sip:asterisk@110.5.42.156:5060>
Call-ID: 5d2369ee192cfe501b1f78520102b18d@110.5.42.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.23.0
Date: Tue, 23 Aug 2016 10:56:02 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=z9hG4bK118080c0;rport=5060
From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as6d1b2a20
To: <sip:200@10.0.1.34:5060>;tag=325588342
Call-ID: 5d2369ee192cfe501b1f78520102b18d@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 '5d2369ee192cfe501b1f78520102b18d@110.5.42.156:5060' Method: OPTIONS
localh