Asterisk SVN-branch-1.4-r118251, Copyright (C) 1999 - 2008 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 SVN-branch-1.4-r118251 currently running on localhost (pid = 5313)
Verbosity was 3 and is now 22
localhost*CLI> sip set debug
SIP Debugging enabled
Really destroying SIP dialog '223a1544-428b-422e-bb6a-32591d64c592' Method: ACK

<--- SIP read from 87.236.144.9:5060 --->
INVITE sip:5678@sero.ahwar.net SIP/2.0
Record-Route: <sip:87.236.144.9;r2=on;lr=on>
Record-Route: <sip:87.236.144.9:443;transport=tls;r2=on;lr=on>
Via: SIP/2.0/UDP 87.236.144.9;branch=z9hG4bK83c5.cdd5bd14.0;i=1
Via: SIP/2.0/TLS 192.168.0.164;received=193.227.186.146;branch=z9hG4bK12XfqevAyU;rport=2363
Call-ID: 4d36eca4-34cf-4ae1-ae58-2ce2bbf3c7d6
CSeq: 41153 INVITE
Max-Forwards: 69
Accept-Encoding: deflate
User-Agent: Malagueta/3.0 (Windows; Windows NT 5.1; IE 6.0.2900.2180)
From: "9613089187" <sip:9613089187@sero.ahwar.net>;tag=6ZygZ-F
To: "5678" <sip:5678@sero.ahwar.net>
Contact: <sip:9613089187@193.227.186.146:2363;transport=UDP;rinstance=D98C1DD404B2008F980980E97E42F8EC>
Content-Type: application/sdp
Content-Length: 819
P-hint: outbound

v=0
o=- 1218637441 878320812 IN IP4 192.168.0.164
s=-
c=IN IP4 192.168.0.164
t=0 0
a=ice-pwd:YfXJ7RU03Ysy4LqXXIvXtJ
a=ice-ufrag:/Agp
m=audio 15104 RTP/AVP 103 18 102 0 8 97 119 117 100 101 13 105 106
a=rtcp:15105
a=candidate:1 1 UDP 2122301695 192.168.0.164 15104
a=candidate:2 1 UDP 1694482431 193.227.186.146 15104
a=candidate:3 1 UDP 16744447 87.236.144.70 42402
a=candidate:1 2 UDP 2122301694 192.168.0.164 15105
a=candidate:2 2 UDP 1694482430 193.227.186.146 15105
a=candidate:3 2 UDP 16744446 87.236.144.70 42403
a=rtpmap:103 ISAC/16000
a=fmtp:18 annexb=no
a=rtpmap:102 iLBC/8000
a=rtpmap:97 IPCMWB/16000
a=rtpmap:119 ISACLC/16000
a=rtpmap:117 red/8000
a=rtpmap:100 EG711U/8000
a=rtpmap:101 EG711A/8000
a=rtpmap:105 CN/16000
a=rtpmap:106 telephone-event/8000
a=fmtp:106 0-16
a=sendrecv

<------------->
--- (16 headers 27 lines) ---
Sending to 87.236.144.9 : 5060 (no NAT)
Using INVITE request as basis request - 4d36eca4-34cf-4ae1-ae58-2ce2bbf3c7d6
Found no matching peer or user for '87.236.144.9:5060'
Found RTP audio format 103
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 119
Found RTP audio format 117
Found RTP audio format 100
Found RTP audio format 101
Found RTP audio format 13
Found RTP audio format 105
Found RTP audio format 106
Peer audio RTP is at port 192.168.0.164:15104
Found unknown media description format ISAC for ID 103
Found audio description format iLBC for ID 102
Found unknown media description format IPCMWB for ID 97
Found unknown media description format ISACLC for ID 119
Found unknown media description format red for ID 117
Found unknown media description format EG711U for ID 100
Found unknown media description format EG711A for ID 101
Found audio description format CN for ID 105
Found audio description format telephone-event for ID 106
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.164:15104
Looking for 5678 in from-sip-external (domain sero.ahwar.net)
list_route: hop: <sip:87.236.144.9;r2=on;lr=on>
list_route: hop: <sip:87.236.144.9:443;transport=tls;r2=on;lr=on>

<--- Transmitting (no NAT) to 87.236.144.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 87.236.144.9;branch=z9hG4bK83c5.cdd5bd14.0;i=1;received=87.236.144.9
Via: SIP/2.0/TLS 192.168.0.164;received=193.227.186.146;branch=z9hG4bK12XfqevAyU;rport=2363
Record-Route: <sip:87.236.144.9;r2=on;lr=on>
Record-Route: <sip:87.236.144.9:443;transport=tls;r2=on;lr=on>
From: "9613089187" <sip:9613089187@sero.ahwar.net>;tag=6ZygZ-F
To: "5678" <sip:5678@sero.ahwar.net>
Call-ID: 4d36eca4-34cf-4ae1-ae58-2ce2bbf3c7d6
CSeq: 41153 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5678@87.236.144.9>
Content-Length: 0

<------------>
    -- Executing [5678@from-sip-external:1] NoOp("SIP/sero.ahwar.net-09972498", "Received incoming SIP connection from unknown peer to 5678") in new stack
    -- Executing [5678@from-sip-external:2] Set("SIP/sero.ahwar.net-09972498", "DID=5678") in new stack
    -- Executing [5678@from-sip-external:3] Goto("SIP/sero.ahwar.net-09972498", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/sero.ahwar.net-09972498", "1?from-trunk|5678|1") in new stack
    -- Goto (from-trunk,5678,1)
Scheduling destruction of SIP dialog '4d36eca4-34cf-4ae1-ae58-2ce2bbf3c7d6' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 87.236.144.9:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 87.236.144.9;branch=z9hG4bK83c5.cdd5bd14.0;i=1;received=87.236.144.9
Via: SIP/2.0/TLS 192.168.0.164;received=193.227.186.146;branch=z9hG4bK12XfqevAyU;rport=2363
From: "9613089187" <sip:9613089187@sero.ahwar.net>;tag=6ZygZ-F
To: "5678" <sip:5678@sero.ahwar.net>;tag=as30537b8b
Call-ID: 4d36eca4-34cf-4ae1-ae58-2ce2bbf3c7d6
CSeq: 41153 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5678@87.236.144.9>
Content-Length: 0

<------------>
localhost*CLI> 
<--- SIP read from 87.236.144.9:5060 --->
ACK sip:5678@sero.ahwar.net SIP/2.0
Via: SIP/2.0/UDP 87.236.144.9;branch=z9hG4bK83c5.cdd5bd14.0
From: "9613089187" <sip:9613089187@sero.ahwar.net>;tag=6ZygZ-F
Call-ID: 4d36eca4-34cf-4ae1-ae58-2ce2bbf3c7d6
To: "5678" <sip:5678@sero.ahwar.net>;tag=as30537b8b
CSeq: 41153 ACK
Max-Forwards: 70
User-Agent: OpenSER (1.3.2-tls (i386/linux))
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (NAT) to 87.236.144.39:5060:
OPTIONS sip:87.236.144.39 SIP/2.0
Via: SIP/2.0/UDP 87.236.144.35:5060;branch=z9hG4bK1c3b08a5;rport
From: "Unknown" <sip:Unknown@87.236.144.35>;tag=as6ed1ab77
To: <sip:87.236.144.39>
Contact: <sip:Unknown@87.236.144.35>
Call-ID: 5ec7b4f84035cb576da0819c32fdcc30@87.236.144.35
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Aug 2008 14:24:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

---
localhost*CLI> 
<--- SIP read from 87.236.144.39:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.236.144.35:5060;branch=z9hG4bK1c3b08a5;rport
From: "Unknown" <sip:Unknown@87.236.144.35>;tag=as6ed1ab77
To: <sip:87.236.144.39>
Call-ID: 5ec7b4f84035cb576da0819c32fdcc30@87.236.144.35
CSeq: 102 OPTIONS
Contact: <sip:87.236.144.39:5060>
User-Agent: Asterisk PBX
Date: Wed, 13 Aug 2008 14:24:43 GMT
Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
Accept: application/sdp, application/qsig, text/plain, application/dtmf-relay
Content-Length: 0