clilog.txt

*CLI> dialplan show rmt-context
[ Context 'rmt-context' created by 'pbx_config' ]
  '4001020' =>      1. Goto(s,1)                                  [pbx_config]
  'failed' =>       1. Noop()                                     [pbx_config]
  'h' =>            1. Noop()                                     [pbx_config]
  's' =>            1. answer()                                   [pbx_config]
                    2. Echo()                                     [pbx_config]
  '_.' =>           1. GoTo(s,1)                                  [pbx_config]

-= 5 extensions (6 priorities) in 1 context. =-
*CLI> sip set debug on
SIP Debugging enabled
*CLI>
<--- SIP read from UDP:192.168.8.1:5062 --->
INVITE sip:4001020@192.168.8.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5062;branch=z9hG4bK7f1a3d33
Max-Forwards: 70
From: "DISP" <sip:RMT20@sop-korniychuk>;tag=as518542c7
To: <sip:4001020@192.168.8.2:5060>
Contact: <sip:RMT20@192.168.8.1:5062>
Call-ID: 6fcce09531cdaa91098e7a146d90056d@sop-korniychuk
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 17 Apr 2012 11:32:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 584847977 584847977 IN IP4 192.168.8.1
s=Asterisk PBX 10.2.0-rc2
c=IN IP4 192.168.8.1
t=0 0
m=audio 15568 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.8.1:5062 (no NAT)
Using INVITE request as basis request - 6fcce09531cdaa91098e7a146d90056d@sop-korniychuk
Found peer 'RMT20' for 'RMT20' from 192.168.8.1:5062

<--- Reliably Transmitting (no NAT) to 192.168.8.1:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.8.1:5062;branch=z9hG4bK7f1a3d33;received=192.168.8.1
From: "DISP" <sip:RMT20@sop-korniychuk>;tag=as518542c7
To: <sip:4001020@192.168.8.2:5060>;tag=as5c04e255
Call-ID: 6fcce09531cdaa91098e7a146d90056d@sop-korniychuk
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ede4fd2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '6fcce09531cdaa91098e7a146d90056d@sop-korniychuk' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.8.1:5062 --->
ACK sip:4001020@192.168.8.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5062;branch=z9hG4bK7f1a3d33
Max-Forwards: 70
From: "DISP" <sip:RMT20@sop-korniychuk>;tag=as518542c7
To: <sip:4001020@192.168.8.2:5060>;tag=as5c04e255
Contact: <sip:RMT20@192.168.8.1:5062>
Call-ID: 6fcce09531cdaa91098e7a146d90056d@sop-korniychuk
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

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

<--- SIP read from UDP:192.168.8.1:5062 --->
INVITE sip:4001020@192.168.8.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5062;branch=z9hG4bK1e12fbed
Max-Forwards: 70
From: "DISP" <sip:RMT20@sop-korniychuk>;tag=as518542c7
To: <sip:4001020@192.168.8.2:5060>
Contact: <sip:RMT20@192.168.8.1:5062>
Call-ID: 6fcce09531cdaa91098e7a146d90056d@sop-korniychuk
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Authorization: Digest username="RMT20", realm="asterisk", algorithm=MD5, uri="sip:4001020@192.168.8.2:5060", nonce="3ede4fd2", response="a3fa9deea96f1820b1e3462ef77072e1"
Date: Tue, 17 Apr 2012 11:32:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 584847977 584847978 IN IP4 192.168.8.1
s=Asterisk PBX 10.2.0-rc2
c=IN IP4 192.168.8.1
t=0 0
m=audio 15568 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 11 lines) ---
Sending to 192.168.8.1:5062 (no NAT)
Using INVITE request as basis request - 6fcce09531cdaa91098e7a146d90056d@sop-korniychuk
Found peer 'RMT20' for 'RMT20' from 192.168.8.1:5062
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.8.1:15568

<--- Reliably Transmitting (no NAT) to 192.168.8.1:5062 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.8.1:5062;branch=z9hG4bK1e12fbed;received=192.168.8.1
From: "DISP" <sip:RMT20@sop-korniychuk>;tag=as518542c7
To: <sip:4001020@192.168.8.2:5060>;tag=as5c04e255
Call-ID: 6fcce09531cdaa91098e7a146d90056d@sop-korniychuk
CSeq: 103 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 17 15:33:08] NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20' (192.168.8.1:5062) to extension '4001020' rejected because extension not found in context 'rmt-context'.
Scheduling destruction of SIP dialog '6fcce09531cdaa91098e7a146d90056d@sop-korniychuk' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.8.1:5062 --->
ACK sip:4001020@192.168.8.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5062;branch=z9hG4bK1e12fbed
Max-Forwards: 70
From: "DISP" <sip:RMT20@sop-korniychuk>;tag=as518542c7
To: <sip:4001020@192.168.8.2:5060>;tag=as5c04e255
Contact: <sip:RMT20@192.168.8.1:5062>
Call-ID: 6fcce09531cdaa91098e7a146d90056d@sop-korniychuk
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0