SIP Debugging re-enabled
bath-office-pbx1*CLI>
<--- SIP read from 193.111.200.11:5082 --->
INVITE sip:101@195.112.25.179 SIP/2.0
To: <sip:101@195.112.25.179>
From: "Keith Sudbury"<sip:2115357@lon-pbx-12.netzensolutions.com>;tag=3875437f
Via: SIP/2.0/UDP 193.111.200.11:5082;branch=z9hG4bK-d87543-35a2da6c6112924d72a3-1-cHBmMmYzNzY0YTZjZWM3NjcwOTk4Nw..-d87543-
Call-ID: 83114487914ab0bf433a1e053dbbc91e
CSeq: 98564564 INVITE
Contact: <sip:987Gez5w2MtBftVTHd2LCYLyzc1d2itPlAlfHLHr7g.@193.111.200.11:5082;line=ydg4eu9f>;flow-id=1
Max-Forwards: 69
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO
Content-Type: application/sdp
Supported: timer, replaces, callerid
User-Agent: snom360/6.5.8
Allow-Events: talk, hold, refer
Content-Length: 453
P-Key-Flags: resolution="31x13", keys="4"
Session-Expires: 3600;refresher=uas
Min-SE: 90
v=0
o=root 48522265 1728620696 IN IP4 193.111.200.11
s=call
c=IN IP4 193.111.200.11
t=0 0
m=audio 10508 RTP/AVP 0 8 18 2 3 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:sybe9dg6cvzOc9nmhfJhkkro/s3XdP2Dxsnqfe1s
a=encryption:optional
a=fmtp:101 0-16
a=ptime:20
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:18 g729/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (18 headers 18 lines) ---
Sending to 193.111.200.11 : 5082 (no NAT)
Using INVITE request as basis request - 83114487914ab0bf433a1e053dbbc91e
Found no matching peer or user for '193.111.200.11:5082'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 193.111.200.11:10508
Found description format pcmu for ID 0
Found description format pcma for ID 8
Found description format g729 for ID 18
Found description format g726-32 for ID 2
Found description format gsm for ID 3
Found description format g723 for ID 4
Found description format telephone-event for ID 101
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 193.111.200.11:10508
Looking for 101 in default (domain 195.112.25.179)
<--- Reliably Transmitting (no NAT) to 193.111.200.11:5082 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 193.111.200.11:5082;branch=z9hG4bK-d87543-35a2da6c6112924d72a3-1-cHBmMmYzNzY0YTZjZWM3NjcwOTk4Nw..-d87543-;received=193.111.200.11
From: "Keith Sudbury"<sip:2115357@lon-pbx-12.netzensolutions.com>;tag=3875437f
To: <sip:101@195.112.25.179>;tag=as4c82a484
Call-ID: 83114487914ab0bf433a1e053dbbc91e
CSeq: 98564564 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Oct 29 16:24:01] NOTICE[9866]: chan_sip.c:13669 handle_request_invite: Call from '' to extension '101' rejected because extension not found.
Scheduling destruction of SIP dialog '83114487914ab0bf433a1e053dbbc91e' in 32000 ms (Method: INVITE)
bath-office-pbx1*CLI>
<--- SIP read from 193.111.200.11:5082 --->
ACK sip:101@195.112.25.179 SIP/2.0
To: <sip:101@195.112.25.179>;tag=as4c82a484
From: "Keith Sudbury"<sip:2115357@lon-pbx-12.netzensolutions.com>;tag=3875437f
Via: SIP/2.0/UDP 193.111.200.11:5082;branch=z9hG4bK-d87543-35a2da6c6112924d72a3-1-cHBmMmYzNzY0YTZjZWM3NjcwOTk4Nw..-d87543-
Call-ID: 83114487914ab0bf433a1e053dbbc91e
CSeq: 98564564 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '83114487914ab0bf433a1e053dbbc91e' Method: ACK