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" <[EMAIL PROTECTED]> To: <asterisk-users@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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]> > Date: Sun, 06 Nov 2005 02:20:59 GMT > Call-ID: [EMAIL PROTECTED] > 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:[EMAIL PROTECTED]>;tag=as5568042c > Call-ID: [EMAIL PROTECTED] > CSeq: 101 INVITE > User-Agent: VBuzzer/1.0 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > 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 > '[EMAIL PROTECTED]' in 15000 ms > > > In the First line Invite for 5505 is the extension I have registered for > Vbuzzer with > username:[EMAIL PROTECTED]:80/5505 > > Why is this happening.................. > Plz help... any one.......... > Plz > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
