For vbuzzer, you must have an extension in the format exten => yourvbuzzerusername,....
in your extensions.conf, in the context associated with your vbuzzer peer. That's where your incoming calls will land I don't know about voipgo, I haven't used them Liviu ----- Original Message ----- From: "Yajie Si" <[email protected]> To: <[email protected]> Sent: Tuesday, February 24, 2009 3:36 PM Subject: [on-asterisk] 404 not found Hi, I am having a problem with my trixbox registered to some ITSPs (namely) vbuzzer and voipgo. outgoing calls are working fine. The box registerated to ITSP's switch OK. CLI "sip show registry" shows "registered". But the incoming calls doesn't work and go to the ITSP's user Voice mail directly. I capatured packet on the trixbox. The packet captures shows when incoming "invite" msg come from ITSP, the box reponses it with "404 not found". This is confirmed by the support of ITSP. One thing i found is the Request_URI from ITSP invite msg is usern...@domain, the username part is my sip user name, not my phone number. for example: [email protected] <[email protected]>, or [email protected], not [email protected]<[email protected]>or 647 [email protected] <[email protected]>. Does trixbox only accept the phone number, not user name in invite message. The below is the capture of vbuzzer session. Does anyone know how to fix it? Thanks! Frame 109 (1328 bytes on wire, 1328 bytes captured) Ethernet II, Src: Cisco-Li_7e:bc:1d (00:1e:e5:7e:bc:1d), Dst: Aastra_11:8f:3b (00:08:5d:11:8f:3b) Internet Protocol, Src: 209.47.41.24 (209.47.41.24), Dst: 192.168.1.103 (192.168.1.103) User Datagram Protocol, Src Port: http (80), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:[email protected]<sip%[email protected]>SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route:<sip:209.47.41.24:80;lr=on;port1=80;port2=80> Record-Route:<sip:209.47.41.48:80;lr=on;port1=80;port2=80> Via:SIP/2.0/UDP 209.47.41.24:80 ;branch=z9hG4bK-3b443a3b4b7227564b4a476d656b3270;port1=80;port2=80 Via:SIP/2.0/UDP 209.47.41.48:80 ;branch=z9hG4bK-442a244d7365627b42345a6f2a632340;port1=80;port2=80 Via: SIP/2.0/UDP 209.47.41.61:5060 ;x-route-tag="tgrp:sroutetor1";branch=z9hG4bK2A2D6B423 From: <sip:[email protected] <sip%[email protected]> >;tag=95D01268-1462 SIP from address: sip:[email protected]<sip%[email protected]> SIP tag: 95D01268-1462 To: <sip:[email protected] <sip%[email protected]>> SIP to address: sip:[email protected]<sip%[email protected]> Call-ID: [email protected] User-Agent: Cisco-SIPGateway/IOS-12.x CSeq: 101 INVITE Sequence Number: 101 Method: INVITE Max-Forwards: 69 Contact: <sip:[email protected]:5060> Contact Binding: <sip:[email protected]:5060> URI: <sip:[email protected]:5060> SIP contact address: sip:[email protected]:5060 Content-Type: application/sdp Content-Length:330 Caller-Ip:209.47.41.61 Remote-Party-ID:<sip:[email protected]<sip%[email protected]> ;user=phone>;party=calling;screen=yes;privacy=off P-Asserted-Identity:<sip:[email protected]<sip%[email protected]> > From-Cluster:Y Message Body Frame 110 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: Aastra_11:8f:3b (00:08:5d:11:8f:3b), Dst: Cisco-Li_7e:bc:1d (00:1e:e5:7e:bc:1d) Internet Protocol, Src: 192.168.1.103 (192.168.1.103), Dst: 209.47.41.24 (209.47.41.24) User Datagram Protocol, Src Port: sip (5060), Dst Port: http (80) Session Initiation Protocol Status-Line: SIP/2.0 404 Not Found Status-Code: 404 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 209.47.41.24:80 ;branch=z9hG4bK-3b443a3b4b7227564b4a476d656b3270;port1=80;port2=80;received=209.47.41.24 Via: SIP/2.0/UDP 209.47.41.48:80 ;branch=z9hG4bK-442a244d7365627b42345a6f2a632340;port1=80;port2=80 Via: SIP/2.0/UDP 209.47.41.61:5060 ;x-route-tag="tgrp:sroutetor1";branch=z9hG4bK2A2D6B423 From: <sip:[email protected] <sip%[email protected]> >;tag=95D01268-1462 SIP from address: sip:[email protected]<sip%[email protected]> SIP tag: 95D01268-1462 To: <sip:[email protected] <sip%[email protected]> >;tag=as34e7e48f SIP to address: sip:[email protected]<sip%[email protected]> SIP tag: as34e7e48f Call-ID: [email protected] CSeq: 101 INVITE Sequence Number: 101 Method: INVITE User-Agent: AastraLinkPro160/1.2int257 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -- Roger Si --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
