CREATE TABLE IF NOT EXISTS `extensions_table` (
  `id` int(11) NOT NULL auto_increment,
  `context` varchar(20) NOT NULL default '',
  `exten` varchar(20) NOT NULL default '',
  `priority` tinyint(4) NOT NULL default '0',
  `app` varchar(20) NOT NULL default '',
  `appdata` varchar(128) NOT NULL default '',
  PRIMARY KEY  (`context`,`exten`,`priority`),
  KEY `id` (`id`)
) ENGINE=MyISAM  DEFAULT CHARSET=latin1 AUTO_INCREMENT=3 ;

--
-- Daten für Tabelle `extensions_table`
--

INSERT INTO `extensions_table` (`id`, `context`, `exten`, `priority`, `app`, `appdata`) VALUES
(1, 'Sip-Out', '_.', 1, 'Set', 'NR=${EXTEN}'),
(2, 'Sip-Out', '_.', 2, 'Wait', '100');



troying SIP dialog '5dbdcb5d1b287b390dc106ae6947a3f8@81.89.105.39' Method: OPTIONS
81-89-105-39*CLI>
<--- SIP read from UDP:79.197.196.90:23710 --->
INVITE sip:222@sogo.dyndns.org SIP/2.0
Via: SIP/2.0/UDP 192.168.178.103:23710;branch=z9hG4bK-d8754z-8f4d687d8dc71476-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:111@79.197.196.90:23710>
To: "222"<sip:222@sogo.dyndns.org>
From: "111"<sip:111@sogo.dyndns.org>;tag=71ef9721
Call-ID: NWEwMmUwYTJjNWQ5NGUyMDE1Nzk0ZGFhYTVlMTlmMGU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1014k stamp 56015
Content-Length: 345

v=0
o=- 2 2 IN IP4 192.168.178.103
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.178.103
t=0 0
m=audio 63500 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=alt:1 3 : 71BqR3rm eptGCRIn 192.168.178.103 63500
a=alt:2 2 : 6Ja8YDVP lbycaeA5 172.16.188.1 63500
a=alt:3 1 : e/hKKwoL 9GGvwtRA 192.168.167.1 63500
a=sendrecv

<------------->
--- (12 headers 12 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 79.197.196.90 : 23710 (NAT)
Using INVITE request as basis request - NWEwMmUwYTJjNWQ5NGUyMDE1Nzk0ZGFhYTVlMTlmMGU.
Reliably Transmitting (NAT) to 79.197.196.90:50156:
OPTIONS sip:79.197.196.90 SIP/2.0
Via: SIP/2.0/UDP 81.89.105.39:5060;branch=z9hG4bK2df87bff;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@81.89.105.39>;tag=as35e252c9
To: <sip:79.197.196.90>
Contact: <sip:asterisk@81.89.105.39>
Call-ID: 7a27a1cf38ef07871fe473a92e2bc81a@81.89.105.39
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Mon, 21 Jun 2010 20:25:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Jun 22 00:25:25] NOTICE[1774]: chan_sip.c:22895 sip_poke_peer: Still have a QUALIFY dialog active, deleting
Really destroying SIP dialog '7a27a1cf38ef07871fe473a92e2bc81a@81.89.105.39' Method: OPTIONS
Reliably Transmitting (NAT) to 79.197.196.90:50156:
OPTIONS sip:111@192.168.178.48:50156;rinstance=3A63A2E61FF2D7B80B4C SIP/2.0
Via: SIP/2.0/UDP 81.89.105.39:5060;branch=z9hG4bK1fcc49b5;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@81.89.105.39>;tag=as5ed30c95
To: <sip:111@192.168.178.48:50156;rinstance=3A63A2E61FF2D7B80B4C>
Contact: <sip:asterisk@81.89.105.39>
Call-ID: 444183cb72b4ad4f4589743d62d03113@81.89.105.39
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Mon, 21 Jun 2010 20:25:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Found peer '111' for '111' from 79.197.196.90:23710
81-89-105-39*CLI>
<--- Reliably Transmitting (NAT) to 79.197.196.90:23710 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.178.103:23710;branch=z9hG4bK-d8754z-8f4d687d8dc71476-1---d8754z-;received=79.197.196.90;rport=23710
From: "111"<sip:111@sogo.dyndns.org>;tag=71ef9721
To: "222"<sip:222@sogo.dyndns.org>;tag=as56962c64
Call-ID: NWEwMmUwYTJjNWQ5NGUyMDE1Nzk0ZGFhYTVlMTlmMGU.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7fd38018"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NWEwMmUwYTJjNWQ5NGUyMDE1Nzk0ZGFhYTVlMTlmMGU.' in 14784 ms (Method: INVITE)
81-89-105-39*CLI>
<--- SIP read from UDP:79.197.196.90:23710 --->
ACK sip:222@sogo.dyndns.org SIP/2.0
Via: SIP/2.0/UDP 192.168.178.103:23710;branch=z9hG4bK-d8754z-8f4d687d8dc71476-1---d8754z-;rport
To: "222"<sip:222@sogo.dyndns.org>;tag=as56962c64
From: "111"<sip:111@sogo.dyndns.org>;tag=71ef9721
Call-ID: NWEwMmUwYTJjNWQ5NGUyMDE1Nzk0ZGFhYTVlMTlmMGU.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
81-89-105-39*CLI>
<--- SIP read from UDP:79.197.196.90:23710 --->
INVITE sip:222@sogo.dyndns.org SIP/2.0
Via: SIP/2.0/UDP 192.168.178.103:23710;branch=z9hG4bK-d8754z-47d5402d60ba356f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:111@79.197.196.90:23710>
To: "222"<sip:222@sogo.dyndns.org>
From: "111"<sip:111@sogo.dyndns.org>;tag=71ef9721
Call-ID: NWEwMmUwYTJjNWQ5NGUyMDE1Nzk0ZGFhYTVlMTlmMGU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1014k stamp 56015
Authorization: Digest username="111",realm="asterisk",nonce="7fd38018",uri="sip:222@sogo.dyndns.org",response="fcbac3a14eeb1b032ca83a6ab782f4d4",algorithm=MD5
Content-Length: 345

v=0
o=- 2 2 IN IP4 192.168.178.103
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.178.103
t=0 0
m=audio 63500 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=alt:1 3 : 71BqR3rm eptGCRIn 192.168.178.103 63500
a=alt:2 2 : 6Ja8YDVP lbycaeA5 172.16.188.1 63500
a=alt:3 1 : e/hKKwoL 9GGvwtRA 192.168.167.1 63500
a=sendrecv

<------------->
--- (13 headers 12 lines) ---
Sending to 79.197.196.90 : 23710 (NAT)
Using INVITE request as basis request - NWEwMmUwYTJjNWQ5NGUyMDE1Nzk0ZGFhYTVlMTlmMGU.
Reliably Transmitting (NAT) to 79.197.196.90:50156:
OPTIONS sip:79.197.196.90 SIP/2.0
Via: SIP/2.0/UDP 81.89.105.39:5060;branch=z9hG4bK155bba40;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@81.89.105.39>;tag=as3041f6d4
To: <sip:79.197.196.90>
Contact: <sip:asterisk@81.89.105.39>
Call-ID: 5c9475654943859459afd76806e9e4b4@81.89.105.39
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Mon, 21 Jun 2010 20:25:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Jun 22 00:25:25] NOTICE[1774]: chan_sip.c:22895 sip_poke_peer: Still have a QUALIFY dialog active, deleting
Really destroying SIP dialog '5c9475654943859459afd76806e9e4b4@81.89.105.39' Method: OPTIONS
Reliably Transmitting (NAT) to 79.197.196.90:50156:
OPTIONS sip:111@192.168.178.48:50156;rinstance=3A63A2E61FF2D7B80B4C SIP/2.0
Via: SIP/2.0/UDP 81.89.105.39:5060;branch=z9hG4bK2bedb322;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@81.89.105.39>;tag=as72354774
To: <sip:111@192.168.178.48:50156;rinstance=3A63A2E61FF2D7B80B4C>
Contact: <sip:asterisk@81.89.105.39>
Call-ID: 1c4d99db22677e7f4005407467e7d0a2@81.89.105.39
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Mon, 21 Jun 2010 20:25:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Found peer '111' for '111' from 79.197.196.90:23710
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.178.103:63500
Looking for 222 in Sip-Out (domain sogo.dyndns.org)
81-89-105-39*CLI>
<--- Reliably Transmitting (NAT) to 79.197.196.90:23710 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.178.103:23710;branch=z9hG4bK-d8754z-47d5402d60ba356f-1---d8754z-;received=79.197.196.90;rport=23710
From: "111"<sip:111@sogo.dyndns.org>;tag=71ef9721
To: "222"<sip:222@sogo.dyndns.org>;tag=as56962c64
Call-ID: NWEwMmUwYTJjNWQ5NGUyMDE1Nzk0ZGFhYTVlMTlmMGU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Jun 22 00:25:25] NOTICE[1774]: chan_sip.c:20059 handle_request_invite: Call from '111' to extension '222' rejected because extension not found.
Scheduling destruction of SIP dialog 'NWEwMmUwYTJjNWQ5NGUyMDE1Nzk0ZGFhYTVlMTlmMGU.' in 14784 ms (Method: INVITE)
81-89-105-39*CLI>
<--- SIP read from UDP:79.197.196.90:23710 --->
ACK sip:222@sogo.dyndns.org SIP/2.0
Via: SIP/2.0/UDP 192.168.178.103:23710;branch=z9hG4bK-d8754z-47d5402d60ba356f-1---d8754z-;rport
To: "222"<sip:222@sogo.dyndns.org>;tag=as56962c64
From: "111"<sip:111@sogo.dyndns.org>;tag=71ef9721
Call-ID: NWEwMmUwYTJjNWQ5NGUyMDE1Nzk0ZGFhYTVlMTlmMGU.
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
81-89-105-39*CLI>
<--- SIP read from UDP:79.197.196.90:50156 --->
SIP/2.0 200 OK
From: "asterisk" <sip:asterisk@81.89.105.39>;tag=as35e252c9
Call-ID: 7a27a1cf38ef07871fe473a92e2bc81a@81.89.105.39
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 81.89.105.39:5060;branch=z9hG4bK2df87bff;rport
To: <sip:79.197.196.90>
Contact: <sip:111@192.168.178.48:50156>
Allow: OPTIONS, INVITE, ACK, CANCEL, BYE
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
81-89-105-39*CLI>
<--- SIP read from UDP:79.197.196.90:50156 --->
SIP/2.0 200 OK
From: "asterisk" <sip:asterisk@81.89.105.39>;tag=as5ed30c95
Call-ID: 444183cb72b4ad4f4589743d62d03113@81.89.105.39
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 81.89.105.39:5060;branch=z9hG4bK1fcc49b5;rport
To: <sip:111@192.168.178.48:50156;rinstance=3A63A2E61FF2D7B80B4C>
Contact: <sip:111@192.168.178.48:50156>
Allow: OPTIONS, INVITE, ACK, CANCEL, BYE
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '444183cb72b4ad4f4589743d62d03113@81.89.105.39' Method: OPTIONS
81-89-105-39*CLI>
<--- SIP read from UDP:79.197.196.90:50156 --->
SIP/2.0 200 OK
From: "asterisk" <sip:asterisk@81.89.105.39>;tag=as3041f6d4
Call-ID: 5c9475654943859459afd76806e9e4b4@81.89.105.39
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 81.89.105.39:5060;branch=z9hG4bK155bba40;rport
To: <sip:79.197.196.90>
Contact: <sip:111@192.168.178.48:50156>
Allow: OPTIONS, INVITE, ACK, CANCEL, BYE
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
81-89-105-39*CLI>
<--- SIP read from UDP:79.197.196.90:50156 --->
SIP/2.0 200 OK
From: "asterisk" <sip:asterisk@81.89.105.39>;tag=as72354774
Call-ID: 1c4d99db22677e7f4005407467e7d0a2@81.89.105.39
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 81.89.105.39:5060;branch=z9hG4bK2bedb322;rport
To: <sip:111@192.168.178.48:50156;rinstance=3A63A2E61FF2D7B80B4C>
Contact: <sip:111@192.168.178.48:50156>
Allow: OPTIONS, INVITE, ACK, CANCEL, BYE
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '1c4d99db22677e7f4005407467e7d0a2@81.89.105.39' Method: OPTIONS