rendered paste bodyConnected to Asterisk 1.4.43 currently running on tornado (pid = 28890)
Verbosity is at least 3
Really destroying SIP dialog '3a3a5eef2c0c95e30afa760d11adeed2@41.221.5.18' Method: OPTIONS
Scheduling destruction of SIP dialog '059e07606000ce5051d14733404d897d@cyberia.org.za' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 41.135.75.17:14922:
NOTIFY sip:901@41.135.75.17:14922;rinstance=0ac53e6bb59a85f9 SIP/2.0
Via: SIP/2.0/UDP 41.86.105.230:5060;branch=z9hG4bK17590daf;rport
From: "Unknown" <sip:Unknown@cyberia.org.za>;tag=as4a663639
To: <sip:901@41.135.75.17:14922;rinstance=0ac53e6bb59a85f9>
Contact: <sip:Unknown@41.86.105.230>
Call-ID: 059e07606000ce5051d14733404d897d@cyberia.org.za
CSeq: 102 NOTIFY
User-Agent: FPBX-2.9.0(1.4.43)
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 89
Messages-Waiting: no
Message-Account: sip:*97@cyberia.org.za
Voice-Message: 0/0 (0/0)
---
<--- SIP read from 41.135.75.17:14922 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 41.86.105.230:5060;branch=z9hG4bK17590daf;rport=5060
Contact: <sip:192.168.1.4:14922>
To: <sip:901@41.135.75.17:14922;rinstance=0ac53e6bb59a85f9>;tag=018a7957
From: "Unknown"<sip:Unknown@cyberia.org.za>;tag=as4a663639
Call-ID: 059e07606000ce5051d14733404d897d@cyberia.org.za
CSeq: 102 NOTIFY
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '059e07606000ce5051d14733404d897d@cyberia.org.za' Method: NOTIFY
<--- SIP read from 41.221.5.18:5060 --->
INVITE sip:s@41.86.105.230 SIP/2.0
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK62d29311;rport
From: "+27823103007" <sip:+27823103007@41.221.5.18>;tag=as4c610d50
To: <sip:s@41.86.105.230>
Contact: <sip:+27823103007@41.221.5.18>
Call-ID: 785da8e76351ae5630a65ca03794f968@41.221.5.18
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 10 Jan 2012 19:53:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 3160 3160 IN IP4 41.221.5.18
s=session
c=IN IP4 41.221.5.18
t=0 0
m=audio 19692 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 41.221.5.18 : 5060 (NAT)
Using INVITE request as basis request - 785da8e76351ae5630a65ca03794f968@41.221.5.18
Found peer 'andrewcolin'
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x3e1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|png|h261|h263|h263p|h264), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 41.221.5.18:19692
Peer doesn't provide video
Looking for s in andrewc (domain 41.86.105.230)
<--- Reliably Transmitting (NAT) to 41.221.5.18:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK62d29311;received=41.221.5.18;rport=5060
From: "+27823103007" <sip:+27823103007@41.221.5.18>;tag=as4c610d50
To: <sip:s@41.86.105.230>;tag=as560b45d3
Call-ID: 785da8e76351ae5630a65ca03794f968@41.221.5.18
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.4.43)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
[Jan 10 21:52:47] NOTICE[28920]: chan_sip.c:15765 handle_request_invite: Call from 'andrewcolin' to extension 's' rejected because extension not found.
Scheduling destruction of SIP dialog '785da8e76351ae5630a65ca03794f968@41.221.5.18' in 6400 ms (Method: INVITE)
<--- SIP read from 41.221.5.18:5060 --->
ACK sip:s@41.86.105.230 SIP/2.0
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK62d29311;rport
From: "+27823103007" <sip:+27823103007@41.221.5.18>;tag=as4c610d50
To: <sip:s@41.86.105.230>;tag=as560b45d3
Contact: <sip:+27823103007@41.221.5.18>
Call-ID: 785da8e76351ae5630a65ca03794f968@41.221.5.18
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '785da8e76351ae5630a65ca03794f968@41.221.5.18' Method: ACK
<--- SIP read from 41.221.5.18:5060 --->
INVITE sip:s@41.86.105.230 SIP/2.0
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK74c2a7ca;rport
From: "+27823103007" <sip:+27823103007@41.221.5.18>;tag=as7135f24b
To: <sip:s@41.86.105.230>
Contact: <sip:+27823103007@41.221.5.18>
Call-ID: 1ea1e93c54ab8e00210c9b505ed7de74@41.221.5.18
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 10 Jan 2012 19:53:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 3160 3160 IN IP4 41.221.5.18
s=session
c=IN IP4 41.221.5.18
t=0 0
m=audio 16556 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 41.221.5.18 : 5060 (NAT)
Using INVITE request as basis request - 1ea1e93c54ab8e00210c9b505ed7de74@41.221.5.18
Found peer 'andrewcolin'
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x3e1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|png|h261|h263|h263p|h264), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 41.221.5.18:16556
Peer doesn't provide video
Looking for s in andrewc (domain 41.86.105.230)
<--- Reliably Transmitting (NAT) to 41.221.5.18:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK74c2a7ca;received=41.221.5.18;rport=5060
From: "+27823103007" <sip:+27823103007@41.221.5.18>;tag=as7135f24b
To: <sip:s@41.86.105.230>;tag=as5a8e219c
Call-ID: 1ea1e93c54ab8e00210c9b505ed7de74@41.221.5.18
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.4.43)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
[Jan 10 21:52:47] NOTICE[28920]: chan_sip.c:15765 handle_request_invite: Call from 'andrewcolin' to extension 's' rejected because extension not found.
Scheduling destruction of SIP dialog '1ea1e93c54ab8e00210c9b505ed7de74@41.221.5.18' in 6400 ms (Method: INVITE)
<--- SIP read from 41.221.5.18:5060 --->
ACK sip:s@41.86.105.230 SIP/2.0
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK74c2a7ca;rport
From: "+27823103007" <sip:+27823103007@41.221.5.18>;tag=as7135f24b
To: <sip:s@41.86.105.230>;tag=as5a8e219c
Contact: <sip:+27823103007@41.221.5.18>
Call-ID: 1ea1e93c54ab8e00210c9b505ed7de74@41.221.5.18
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1ea1e93c54ab8e00210c9b505ed7de74@41.221.5.18' Method: ACK
tornado*CLI>
Disconnected from Asterisk server
[root@tornado asterisk]#