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

[Alex Balashov]

Try throw the following options into your sip.conf peer:

   port=5060
   insecure=invite,port

Phibee Network Operation Center wrote:

> 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
> 
> 
> 
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-- 
Alex Balashov - Principal
Evariste Systems
Web     : http://www.evaristesys.com/
Tel     : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671