[root@starface ~]# asterisk -rvvvvvvv
Asterisk 11.22.0, 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 11.22.0 currently running on starface (pid = 25725)
starface*CLI> sip set debug on
SIP Debugging enabled
Really destroying SIP dialog '1284917090@127.0.0.1' Method: OPTIONS
<--- SIP read from UDP:192.168.xxx.xxx:5060 --->
INVITE sip:23381023@192.168.xxx.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK6888876465F635109684338017323387;rport
From: <sip:0021123456789@192.168.xxx.xxx;user=phone>;tag=B22A876465F635109683338017323387
To: <sip:723@192.168.xxx.xxx>
Call-ID: 4E2A876465F635109682338017323387
CSeq: 1 INVITE
Contact: <sip:723@192.168.xxx.xxx:5060;transport=udp>
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: Digitalisierungsbox Premium V.10.1.7.124 IPv6, IPSec, PBX
Alert-Info: <http://127.0.0.1>;info=alert-external
Allow-Events: refer, message-summary, dialog
P-Early-Media: supported
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 287
v=0
o=- 30100 1 IN IP4 192.168.xxx.xxx
s=SIP call
c=IN IP4 192.168.xxx.xxx
t=0 0
m=audio 10440 RTP/AVP 8 0 9 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (17 headers 14 lines) ---
Sending to 192.168.xxx.xxx:5060 (no NAT)
Sending to 192.168.xxx.xxx:5060 (no NAT)
Using INVITE request as basis request - 4E2A876465F635109682338017323387
No matching peer for '0021123456789' from '192.168.xxx.xxx:5060'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G722 for ID 9
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|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 192.168.xxx.xxx:10440
Looking for 23381023 in default (domain 192.168.xxx.111)
<--- Reliably Transmitting (no NAT) to 192.168.xxx.xxx:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK6888876465F635109684338017323387;received=192.168.xxx.xxx;rport=5060
From: <sip:0021123456789@192.168.xxx.xxx;user=phone>;tag=B22A876465F635109683338017323387
To: <sip:723@192.168.xxx.xxx>;tag=as16251d66
Call-ID: 4E2A876465F635109682338017323387
CSeq: 1 INVITE
Server: STARFACE PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '4E2A876465F635109682338017323387' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.xxx.xxx:5060 --->
ACK sip:23381023@192.168.xxx.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK6888876465F635109684338017323387;rport
From: <sip:0021123456789@192.168.xxx.xxx;user=phone>;tag=B22A876465F635109683338017323387
To: <sip:723@192.168.xxx.xxx>;tag=as16251d66
Call-ID: 4E2A876465F635109682338017323387
CSeq: 1 ACK
Contact: <sip:723@192.168.xxx.xxx:5060;transport=udp>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4E2A876465F635109682338017323387' Method: ACK
<--- SIP read from UDP:192.168.xxx.10:5060 --->
REGISTER sip:192.168.xxx.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.10:5060;branch=z9hG4bK549083df6421c6812811949c7da63d1;rport
From: <sip:1011.N510IP@192.168.xxx.111>;tag=1276633265
To: <sip:1011.N510IP@192.168.xxx.111>
Call-ID: 147096815@192_168_101_10
CSeq: 92982 REGISTER
Contact: <sip:1011.N510IP@192.168.xxx.10:5060>
Max-Forwards: 70
User-Agent: N510 IP PRO/42.231.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.xxx.10:5060 (no NAT)
Sending to 192.168.xxx.10:5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.xxx.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.xxx.10:5060;branch=z9hG4bK549083df6421c6812811949c7da63d1;received=192.168.xxx.10;rport=5060
From: <sip:1011.N510IP@192.168.xxx.111>;tag=1276633265
To: <sip:1011.N510IP@192.168.xxx.111>;tag=as53547eb0
Call-ID: 147096815@192_168_101_10
CSeq: 92982 REGISTER
Server: STARFACE PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="STARFACE", nonce="0c2c9ac9"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '147096815@192_168_101_10' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.xxx.10:5060 --->
REGISTER sip:192.168.xxx.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.10:5060;branch=z9hG4bK25898c73c1ceb743977ce73c105ec4d0;rport
From: <sip:1011.N510IP@192.168.xxx.111>;tag=1276633265
To: <sip:1011.N510IP@192.168.xxx.111>
Call-ID: 147096815@192_168_101_10
CSeq: 92983 REGISTER
Contact: <sip:1011.N510IP@192.168.xxx.10:5060>
Authorization: Digest username="1011.N510IP", realm="STARFACE", algorithm=MD5, uri="sip:192.168.xxx.111", nonce="0c2c9ac9", response="d034894b2694b696ba5bd58eba410904"
Max-Forwards: 70
User-Agent: N510 IP PRO/42.231.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.xxx.10:5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.xxx.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.xxx.10:5060;branch=z9hG4bK25898c73c1ceb743977ce73c105ec4d0;received=192.168.xxx.10;rport=5060
From: <sip:1011.N510IP@192.168.xxx.111>;tag=1276633265
To: <sip:1011.N510IP@192.168.xxx.111>;tag=as53547eb0
Call-ID: 147096815@192_168_101_10
CSeq: 92983 REGISTER
Server: STARFACE PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 180
Contact: <sip:1011.N510IP@192.168.xxx.10:5060>;expires=180
Date: Fri, 16 Feb 2018 07:51:31 GMT
Content-Length: 0
<------------>
Reliably Transmitting (no NAT) to 192.168.xxx.10:5060:
NOTIFY sip:1011.N510IP@192.168.xxx.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.111:5060;branch=z9hG4bK2127c717;rport
Max-Forwards: 70
Route: <sip:1011.N510IP@192.168.xxx.10:5060>
From: "STARFACE" <sip:STARFACE@192.168.xxx.111>;tag=as6b9f4a7c
To: <sip:1011.N510IP@192.168.xxx.10:5060>;tag=494254289
Contact: <sip:STARFACE@192.168.xxx.111:5060>
Call-ID: 2516398169@192_168_101_10
CSeq: 1370 NOTIFY
User-Agent: STARFACE PBX
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 92
Messages-Waiting: no
Message-Account: sip:pbxvm@192.168.xxx.111
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog '147096815@192_168_101_10' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.xxx.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.xxx.111:5060;branch=z9hG4bK2127c717;rport=5060
From: "STARFACE" <sip:STARFACE@192.168.xxx.111>;tag=as6b9f4a7c
To: <sip:1011.N510IP@192.168.xxx.10:5060>;tag=494254289
Call-ID: 2516398169@192_168_101_10
CSeq: 1370 NOTIFY
User-Agent: N510 IP PRO/42.231.00.000.000
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.xxx.xxx:5060 --->
INVITE sip:23381023@192.168.xxx.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK022E216865F635109689338017323387;rport
From: <sip:0021123456789@192.168.xxx.xxx;user=phone>;tag=4CD0206865F635109688338017323387
To: <sip:723@192.168.xxx.xxx>
Call-ID: E8CF206865F635109687338017323387
CSeq: 1 INVITE
Contact: <sip:723@192.168.xxx.xxx:5060;transport=udp>
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: Digitalisierungsbox Premium V.10.1.7.124 IPv6, IPSec, PBX
Alert-Info: <http://127.0.0.1>;info=alert-external
Allow-Events: refer, message-summary, dialog
P-Early-Media: supported
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 304
v=0
o=- 16877 1 IN IP4 192.168.xxx.xxx
s=SIP call
c=IN IP4 192.168.xxx.xxx
t=0 0
m=audio 10444 RTP/AVP 8 0 9 18 100
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=ptime:20
a=sendrecv
<------------->
--- (17 headers 15 lines) ---
Sending to 192.168.xxx.xxx:5060 (no NAT)
Sending to 192.168.xxx.xxx:5060 (no NAT)
Using INVITE request as basis request - E8CF206865F635109687338017323387
No matching peer for '0021123456789' from '192.168.xxx.xxx:5060'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 100
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G722 for ID 9
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 100
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|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 192.168.xxx.xxx:10444
Looking for 23381023 in default (domain 192.168.xxx.111)
<--- Reliably Transmitting (no NAT) to 192.168.xxx.xxx:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK022E216865F635109689338017323387;received=192.168.xxx.xxx;rport=5060
From: <sip:0021123456789@192.168.xxx.xxx;user=phone>;tag=4CD0206865F635109688338017323387
To: <sip:723@192.168.xxx.xxx>;tag=as6b958255
Call-ID: E8CF206865F635109687338017323387
CSeq: 1 INVITE
Server: STARFACE PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'E8CF206865F635109687338017323387' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.xxx.xxx:5060 --->
ACK sip:23381023@192.168.xxx.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK022E216865F635109689338017323387;rport
From: <sip:0021123456789@192.168.xxx.xxx;user=phone>;tag=4CD0206865F635109688338017323387
To: <sip:723@192.168.xxx.xxx>;tag=as6b958255
Call-ID: E8CF206865F635109687338017323387
CSeq: 1 ACK
Contact: <sip:723@192.168.xxx.xxx:5060;transport=udp>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'E8CF206865F635109687338017323387' Method: ACK
starface*CLI> sip set debug off
SIP Debugging Disabled
-- Remote UNIX connection
-- Remote UNIX connection disconnected
starface*CLI>