this is how it is looking on the console (ip’s and phone numbers obfuscated a bit):
<--- SIP read from 69.XX.XX.21:63647 --->
INVITE sip:91250XXXXXXX@69.XX.XX.23:5060 SIP/2.0
Via: SIP/2.0/UDP 69.XX.XX.21:5060;branch=z9hG4bKA1775
Remote-Party-ID: "Beta" <sip:69.XX.XX.21>;party=calling;screen=no;privacy=off
From: "unknown" <sip:69.XX.XX.23>;tag=1BF378-CC0
To: <sip:91250XXXXXXX@69.XX.XX.23>
Date: Tue, 13 May 2008 14:55:34 GMT
Call-ID: 79733A2F-203311DD-80618575-A42EC5B3@69.XX.XX.21
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1994434739-540217821-2153547125-2754528691
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1210690534
Contact: <sip:69.XX.XX.21:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 188
v=0
o=CiscoSystemsSIP-GW-UserAgent 8246 7594 IN IP4 69.XX.XX.21
s=SIP Call
c=IN IP4 69.XX.XX.21
t=0 0
m=audio 18410 RTP/AVP 0
c=IN IP4 69.XX.XX.21
a=rtpmap:0 PCMU/8000
a=ptime:20
<------------->
--- (21 headers 9 lines) ---
Sending to 69.XX.XX.21 : 5060 (no NAT)
Using INVITE request as basis request - 79733A2F-203311DD-80618575-A42EC5B3@69.XX.XX.21
Found no matching peer or user for '69.XX.XX.21:63647'
Found RTP audio format 0
Peer audio RTP is at port 69.XX.XX.21:18410
Found audio description format PCMU for ID 0
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 69.XX.XX.21:18410
Looking for 91250XXXXXXX in local (domain 69.XX.XX.23)
<--- Reliably Transmitting (no NAT) to 69.XX.XX.21:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 69.XX.XX.21:5060;branch=z9hG4bKA1775;received=69.XX.XX.21
From: "unknown" <sip:69.XX.XX.23>;tag=1BF378-CC0
To: <sip:91250XXXXXXX@69.XX.XX.23>;tag=as7d40bb8e
Call-ID: 79733A2F-203311DD-80618575-A42EC5B3@69.XX.XX.21
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[May 13 14:54:38] NOTICE[3407]: chan_sip.c:13885 handle_request_invite: Call from '' to extension '91250XXXXXXX' rejected because extension not found.
Scheduling destruction of SIP dialog '79733A2F-203311DD-80618575-A42EC5B3@69.XX.XX.21' in 32000 ms (Method: INVITE)
localhost*CLI>
<--- SIP read from 69.XX.XX.21:5060 --->
ACK sip:91250XXXXXXX@69.XX.XX.23:5060 SIP/2.0
Via: SIP/2.0/UDP 69.XX.XX.21:5060;branch=z9hG4bKA1775
From: "unknown" <sip:69.XX.XX.23>;tag=1BF378-CC0
To: <sip:91250XXXXXXX@69.XX.XX.23>;tag=as7d40bb8e
Date: Tue, 13 May 2008 14:55:34 GMT
Call-ID: 79733A2F-203311DD-80618575-A42EC5B3@69.XX.XX.21
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
in comparison to a softphone client that is able to make these calls; this line looks funny
From: "unknown" <sip:69.XX.XX.23>;tag=1BF378-CC0
and this line looks funny
Looking for 91250XXXXXXX in local (domain 69.XX.XX.23)
where on a call that works these lines appear more like:
From: local <sip:6201@69.XX.XX.23>;tag=3783457115
and
Looking for 91250XXXXXXX in numberplan-custom-1 (domain asterisk)
help~!