[Asterisk-Users] Inbound Calls on Asterisk from VBuzzer

[Hitesh Sharma]

Hi

Any one got Inbound Calls from VBuzzer working on Asterisk
I am tried it hard and will be bald in few hours....
The Call comes in... But Gets a 407 Authentication Required from Asterisk
Here is the SIP Log


****************************************************************
Call Comes in from VBuzzer
****************************************************************
Sip read:
INVITE sip:5505 at 24.76.253.179:5060 SIP/2.0
Record-Route: <sip:209.47.41.48:80;ftag=CAFB5090-B12;lr=on>
Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bKfeb5.f3240612.0
Via: SIP/2.0/UDP
209.47.41.61:5060;rport=51854;x-route-tag="tgrp:sroutetor1";branch=z9hG4bK1B
33925D9
From: <sip:209.47.41.61>;tag=CAFB5090-B12
To: <sip:14162733742 at 209.47.41.48>
Date: Sun, 06 Nov 2005 02:20:59 GMT
Call-ID: CDA0996B-4DA211DA-975CC727-E0F535F0 at 209.47.41.61
Supported: timer
Min-SE:  1800
Cisco-Guid: 3449774211-1302467034-3204317201-2459445924
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 4
Timestamp: 1131243659
Contact: <sip:209.47.41.61:51854>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 369
hint: NAThelper
hint: SDP rewritten
hint: usrloc applied
hint: NAT...

v=0
o=CiscoSystemsSIP-GW-UserAgent 3784 2473 IN IP4 209.47.41.61
s=SIP Call
c=IN IP4 209.47.41.61
t=0 0
m=audio 54148 RTP/AVP 0 8 18 3 101
c=IN IP4 209.47.41.27
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive
a=nortpproxy:yes

25 headers, 16 lines
Using latest request as basis request
Sending to 209.47.41.48 : 80 (non-NAT)
Found peer 'vbuzzer'
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required        ***************************
This is what Happens.******************************
Via: SIP/2.0/UDP
209.47.41.48:80;branch=z9hG4bKfeb5.f3240612.0;received=209.47.41.48;rport=80
Via: SIP/2.0/UDP
209.47.41.61:5060;x-route-tag="tgrp:sroutetor1";branch=z9hG4bK1B33925D9
From: <sip:209.47.41.61>;tag=CAFB5090-B12
To: <sip:14162733742 at 209.47.41.48>;tag=as5568042c
Call-ID: CDA0996B-4DA211DA-975CC727-E0F535F0 at 209.47.41.61
CSeq: 101 INVITE
User-Agent: VBuzzer/1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5505 at 24.76.253.179>
Proxy-Authenticate: Digest realm="asterisk", nonce="78594498"
Content-Length: 0


 to 209.47.41.48:80
       > sip_xmit:  0x814ae74 (len 592) to 209.47.41.48 sent via outbound
proxy
       > >>>  Sending SIP message to 209.47.41.48
Scheduling destruction of call
'CDA0996B-4DA211DA-975CC727-E0F535F0 at 209.47.41.61' in 15000 ms


In the First line Invite for 5505 is the extension I have registered for
Vbuzzer with
username:pwd at vbuzzer.com:80/5505

Why is this happening..................
Plz help... any one..........
Plz