[asterisk-users] Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)

[Phibee Network Operation Center]

Phibee Network Operation Center a écrit :
> Hi
>
> Now, my Cisco AS5300 sent call to my asterisk, but two problems:
>
> When i call the phone number, i have:
>
> [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: 
> Call from '' to extension '0426000000' rejected because extension not found.
> [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: 
> Call from '' to extension '0426000000' rejected because extension not found.
>
> (0426000000 = my phone number)
> <..>
>   

I have put a debug:

[Kvoip*CLI>
<--- SIP read from UDP://192.168.50.125:59124 --->
INVITE sip:0426000000 at 192.168.50.130:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.50.125:5060
From: <sip:477000000 at 192.168.50.125>;tag=6950F0-25C7
To: <sip:0426000000 at 192.168.50.130>
Date: Wed, 28 Oct 2009 05:16:26 GMT
Call-ID: E02F04A1-C2B711DE-82B09FC7-B045D36F at 192.168.50.125
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 3761097657-3266777566-2192416711-2957366127
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: 
<sip:477000000 at 192.168.50.125>;party=calling;screen=yes;privacy=off
Timestamp: 1256706986
Contact: <sip:477000000 at 192.168.50.125:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 8642 2741 IN IP4 192.168.50.125
s=SIP Call
c=IN IP4 192.168.50.125
t=0 0
m=audio 18726 RTP/AVP 8 101
c=IN IP4 192.168.50.125
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
[Kvoip*CLI> --- (20 headers 11 lines) ---
[Kvoip*CLI> Sending to 192.168.50.125 : 5060 (no NAT)
[Kvoip*CLI> Using INVITE request as basis request - 
E02F04A1-C2B711DE-82B09FC7-B045D36F at 192.168.50.125
[Kvoip*CLI> No matching peer for '477000000' from '192.168.50.125:59124'
[Kvoip*CLI> Found RTP audio format 8
[Kvoip*CLI> Found RTP audio format 101
[Kvoip*CLI> Peer audio RTP is at port 192.168.50.125:18726
[Kvoip*CLI> Found audio description format PCMA for ID 8
[Kvoip*CLI> Found audio description format telephone-event for ID 101
[Kvoip*CLI> Got unsupported a:fmtp in SDP offer
[Kvoip*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - 
audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 
(alaw)
[Kvoip*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), 
peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Kvoip*CLI> Peer audio RTP is at port 192.168.50.125:18726
[Kvoip*CLI> Looking for 0426000000 in default (domain 192.168.50.130)
[Kvoip*CLI> <--- Reliably Transmitting (no NAT) to 192.168.50.125:5060 --->
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP  192.168.50.125:5060;received=192.168.50.125
From: <sip:477000000 at 192.168.50.125>;tag=6950F0-25C7
To: <sip:0426000000 at 192.168.50.130>;tag=as25696e60
Call-ID: E02F04A1-C2B711DE-82B09FC7-B045D36F at 192.168.50.125
CSeq: 101 INVITE

Server: Asterisk PBX 1.6.1.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

Ok, i see that:

    1- Cisco sent the phone number of the caller (477000000)
    2- I have a "To: <sip:0426000000 at 192.168.50.130>"
       192.168.50.130 = My Asterisk Server
       192.168.50.125 = My Cisco AS5300
    3- i have a "No matching peer for '477000000' from 
'192.168.50.125:59124'"
       why he search a peer with "477000000" ??

bye
Jerome