[Asterisk-Users] Fw: Inbound Calls on Asterisk from VBuzzer

[Hitesh Sharma]

To Add Here is the VBuzzer Peer Definition

[vbuzzer]
type=peer              ; we only want to call out, not be called
context=default
port=80
username=<>
secret=<>
host=vbuzzer.com
fromuser=<>
fromdomain=vbuzzer.com
disallow=all
allow=ulaw
allow=alaw
allow=g729
User-agent=vbuzzer/1.0
Insecure=yes
nat=yes

I have tried with
Peer=friend, peer=user,peer=type
canreinvite=yes,
every thing

:-(

Hitesh Sharma
----- Original Message ----- 
From: "Hitesh Sharma" <hmadra at gmail.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, November 05, 2005 8:23 PM
Subject: Inbound Calls on Asterisk from VBuzzer


> 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
>