Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 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.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': == Found
== Parsing '/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux5 (pid = 5842)
Verbosity is at least 3
linux5*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2101 (Unspecified) D 5060 Unmonitored
2102 (Unspecified) D 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux5*CLI> sip set debug off
SIP Debugging Disabled
linux5*CLI> sip set debug on
SIP Debugging enabled
linux5*CLI> sip set debug on
SIP Debugging re-enabled
linux5*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2101 (Unspecified) D 5060 Unmonitored
2102 (Unspecified) D 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
<--- SIP read from UDP:192.168.105.2:5060 --->
REGISTER sip:192.168.105.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a86902000000174dff48af000024ec00000000;rport
From: "unknown" <sip:Juzef@192.168.105.1>;tag=3f8417ead17
To: <sip:Juzef@192.168.105.1>
Contact: <sip:Juzef@192.168.105.2>
Call-ID: 893750322122490EB0B48800DEABC4DB0xc0a86902
CSeq: 1 REGISTER
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.105.2 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.105.2:5060 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a86902000000174dff48af000024ec00000000;received=192.168.105.2;rport=5060
From: "unknown" <sip:Juzef@192.168.105.1>;tag=3f8417ead17
To: <sip:Juzef@192.168.105.1>;tag=as12517e42
Call-ID: 893750322122490EB0B48800DEABC4DB0xc0a86902
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.2.9-2+squeeze2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Jun 20 16:18:35] NOTICE[5867]: chan_sip.c:21768 handle_request_register: Registration from '<sip:Juzef@192.168.105.1>' failed for '192.168.105.2' - No matching peer found
Scheduling destruction of SIP dialog '893750322122490EB0B48800DEABC4DB0xc0a86902' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.105.2:5060 --->
REGISTER sip:192.168.105.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a869020000001b4dff48c4000016a000000002;rport
From: "unknown" <sip:2102@192.168.105.1>;tag=3fc817f0017
To: <sip:2102@192.168.105.1>
Contact: <sip:2102@192.168.105.2>
Call-ID: BF7888158C2F42CDB0C984DDBF9364640xc0a86902
CSeq: 1 REGISTER
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.105.2 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.105.2:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a869020000001b4dff48c4000016a000000002;received=192.168.105.2;rport=5060
From: "unknown" <sip:2102@192.168.105.1>;tag=3fc817f0017
To: <sip:2102@192.168.105.1>;tag=as4deb8c14
Call-ID: BF7888158C2F42CDB0C984DDBF9364640xc0a86902
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.2.9-2+squeeze2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0da0bc90"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'BF7888158C2F42CDB0C984DDBF9364640xc0a86902' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.105.2:5060 --->
REGISTER sip:192.168.105.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a869020000001c4dff48c4000026ab00000004;rport
From: "unknown" <sip:2102@192.168.105.1>;tag=3fc817f0017
To: <sip:2102@192.168.105.1>
Contact: <sip:2102@192.168.105.2>
Call-ID: BF7888158C2F42CDB0C984DDBF9364640xc0a86902
CSeq: 2 REGISTER
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0
Authorization: Digest username="2102",realm="asterisk",nonce="0da0bc90",uri="sip:192.168.105.1",response="c054c6abbc5f694e53e6c0d1f7414eb0",algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.105.2 : 5060 (no NAT)
-- Registered SIP '2102' at 192.168.105.2 port 5060
<--- Transmitting (no NAT) to 192.168.105.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a869020000001c4dff48c4000026ab00000004;received=192.168.105.2;rport=5060
From: "unknown" <sip:2102@192.168.105.1>;tag=3fc817f0017
To: <sip:2102@192.168.105.1>;tag=as4deb8c14
Call-ID: BF7888158C2F42CDB0C984DDBF9364640xc0a86902
CSeq: 2 REGISTER
Server: Asterisk PBX 1.6.2.9-2+squeeze2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: <sip:2102@192.168.105.2>;expires=120
Date: Mon, 20 Jun 2011 13:18:56 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'BF7888158C2F42CDB0C984DDBF9364640xc0a86902' in 32000 ms (Method: REGISTER)
linux5*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2101 (Unspecified) D 5060 Unmonitored
2102/2102 192.168.105.2 D 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
Really destroying SIP dialog '893750322122490EB0B48800DEABC4DB0xc0a86902' Method: REGISTER
<--- SIP read from UDP:192.168.105.2:5060 --->
<------------->
Really destroying SIP dialog 'BF7888158C2F42CDB0C984DDBF9364640xc0a86902' Method: REGISTER
<--- SIP read from UDP:192.168.105.2:5060 --->
INVITE sip:2101@192.168.105.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a86902000000224dff48eb0000458000000006;rport
From: "unknown" <sip:2102@192.168.105.1>;tag=2a917f9aa1
To: <sip:2101@192.168.105.1>
Contact: <sip:2102@192.168.105.2>
Call-ID: A0198F840FED4607A13E67027F0C60100xc0a86902
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 368
Content-Type: application/sdp
Supported: replaces,norefersub,timer
v=0
o=- 3517564778 3517564778 IN IP4 192.168.105.2
s=SJphone
c=IN IP4 192.168.105.2
t=0 0
m=audio 49152 RTP/AVP 3 97 98 8 0 101
c=IN IP4 192.168.105.2
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv
<------------->
--- (12 headers 17 lines) ---
== Using SIP RTP CoS mark 5
Sending to 192.168.105.2 : 5060 (no NAT)
Using INVITE request as basis request - A0198F840FED4607A13E67027F0C60100xc0a86902
Found peer '2102' for '2102' from 192.168.105.2:5060
<--- Reliably Transmitting (no NAT) to 192.168.105.2:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a86902000000224dff48eb0000458000000006;received=192.168.105.2;rport=5060
From: "unknown" <sip:2102@192.168.105.1>;tag=2a917f9aa1
To: <sip:2101@192.168.105.1>;tag=as2bbc64c6
Call-ID: A0198F840FED4607A13E67027F0C60100xc0a86902
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.9-2+squeeze2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58bf55a3"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'A0198F840FED4607A13E67027F0C60100xc0a86902' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.105.2:5060 --->
ACK sip:2101@192.168.105.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a86902000000224dff48eb0000458000000006;rport
From: "unknown" <sip:2102@192.168.105.1>;tag=2a917f9aa1
To: <sip:2101@192.168.105.1>;tag=as2bbc64c6
Call-ID: A0198F840FED4607A13E67027F0C60100xc0a86902
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:192.168.105.2:5060 --->
INVITE sip:2101@192.168.105.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a86902000000234dff48ec00002be500000008;rport
From: "unknown" <sip:2102@192.168.105.1>;tag=2a917f9aa1
To: <sip:2101@192.168.105.1>
Contact: <sip:2102@192.168.105.2>
Call-ID: A0198F840FED4607A13E67027F0C60100xc0a86902
CSeq: 2 INVITE
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 368
Content-Type: application/sdp
Supported: replaces,norefersub,timer
Authorization: Digest username="2102",realm="asterisk",nonce="58bf55a3",uri="sip:2101@192.168.105.1",response="70daa3b06aa9a29f4af8f7526b6f40e5",algorithm=MD5
v=0
o=- 3517564778 3517564778 IN IP4 192.168.105.2
s=SJphone
c=IN IP4 192.168.105.2
t=0 0
m=audio 49152 RTP/AVP 3 97 98 8 0 101
c=IN IP4 192.168.105.2
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv
<------------->
--- (13 headers 17 lines) ---
Sending to 192.168.105.2 : 5060 (no NAT)
Using INVITE request as basis request - A0198F840FED4607A13E67027F0C60100xc0a86902
Found peer '2102' for '2102' from 192.168.105.2:5060
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
ound audio description format iLBC for ID 98
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x507 (g723|gsm|ulaw|g729|ilbc), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x406 (gsm|ulaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.105.2:49152
Looking for 2101 in default (domain 192.168.105.1)
<--- Reliably Transmitting (no NAT) to 192.168.105.2:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a86902000000234dff48ec00002be500000008;received=192.168.105.2;rport=5060
From: "unknown" <sip:2102@192.168.105.1>;tag=2a917f9aa1
To: <sip:2101@192.168.105.1>;tag=as2bbc64c6
Call-ID: A0198F840FED4607A13E67027F0C60100xc0a86902
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.9-2+squeeze2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Jun 20 16:19:36] NOTICE[5867]: chan_sip.c:20281 handle_request_invite: Call from '2102' to extension '2101' rejected because extension not found in context 'default'.
Scheduling destruction of SIP dialog 'A0198F840FED4607A13E67027F0C60100xc0a86902' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.105.2:5060 --->
ACK sip:2101@192.168.105.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.105.2;branch=z9hG4bKc0a86902000000234dff48ec00002be500000008;rport
From: "unknown" <sip:2102@192.168.105.1>;tag=2a917f9aa1
To: <sip:2101@192.168.105.1>;tag=as2bbc64c6
Call-ID: A0198F840FED4607A13E67027F0C60100xc0a86902
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0
Authorization: Digest username="2102",realm="asterisk",nonce="58bf55a3",uri="sip:2101@192.168.105.1",response="70daa3b06aa9a29f4af8f7526b6f40e5",algorithm=MD5
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:192.168.105.2:5060 --->
<------------->
<--- SIP read from UDP:192.168.105.2:5060 --->
<------------->
Really destroying SIP dialog 'A0198F840FED4607A13E67027F0C60100xc0a86902' Method: ACK
linux5*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2101 (Unspecified) D 5060 Unmonitored
2102/2102 192.168.105.2 D 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
-- Registered SIP '2102' at 192.168.105.2 port 5060
linux5*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2101 (Unspecified) D 5060 Unmonitored
2102/2102 192.168.105.2 D 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux5*CLI> sip set debug on
SIP Debugging enabled
linux5*CLI> sip set debug off
SIP Debugging Disabled
linux5*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2101 (Unspecified) D 5060 Unmonitored
2102/2102 192.168.105.2 D 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
== Using SIP RTP CoS mark 5
[Jun 20 16:39:27] NOTICE[5867]: chan_sip.c:20281 handle_request_invite: Call from '2102' to extension '2101' rejected because extension not found in context 'default'.
== Using SIP RTP CoS mark 5
[Jun 20 16:40:43] NOTICE[5867]: chan_sip.c:20281 handle_request_invite: Call from '2102' to extension '8000' rejected because extension not found in context 'default'.
linux5*CLI> quit
Executing last minute cleanups