[Serusers] Fw: 400 Bad Request after an ACK

[Rosario Pingaro]

In fact I disabled re-invite in asterisk and everything worked.

Regards
Rosario


----- Original Message ----- 
From: "Jan Bollen" <jbollen at cisco.com>
To: "Rosario Pingaro" <rpingar at nesec.it>
Cc: <serusers at lists.iptel.org>
Sent: Thursday, April 20, 2006 6:30 AM
Subject: Re: [Serusers] Fw: 400 Bad Request after an ACK


>
> Hello Rosario,
>
> Tha ACK contains a branch id = 0 in the first Via: field. I can
> imagine that some implementations are not happy with that.
>
> Via: SIP/2.0/UDP 19x.6x.19x.4x;branch=0
>
> When you're compliant with RFC3261 this branch id. should start with
> "z9hG4bK". The branch id. is used to detect loops.
>
> Best regards,
> Jan
>
>
> On Wed, 19 Apr 2006 Rosario Pingaro kneaded and moulded:
>
> |
> |Can someone help me to debug my problem?
> |
> |I have ser between asterisk and my clients.
> |When I try to call a sip client, it is going to ring. But asap the callee 
> pickup the phone the call goes down.
> |
> |Doing a logging on port 5060 i see that after the ack i get 400 bad 
> request from the sip client.
> |
> |This is the trace:
> |
> |Session Initiation Protocol
> |    Request-Line: ACK sip:0681140017 at 83.211.248.158:62746 SIP/2.0
> |        Method: ACK
> |        Resent Packet: False
> |    Message Header
> |        Record-Route: <sip:19x.6x.19x.4x;ftag=as6d07dd0a;lr=on>
> |        Via: SIP/2.0/UDP 19x.6x.19x.4x;branch=0
> |        Via: SIP/2.0/UDP 
> 19x.24x.16x.9x:5060;branch=z9hG4bK368ed369;rport=5060
> |        From: "anonymous" 
> <sip:asterisk at voip.convergenze.it>;tag=as6d07dd0a
> |            SIP Display info: "anonymous"
> |            SIP from address: sip:asterisk at voip.convergenze.it
> |            SIP tag: as6d07dd0a
> |        To: <sip:08281895109 at voip.convergenze.it>;tag=b1385811e50f0aai1
> |            SIP to address: sip:08281895109 at voip.convergenze.it
> |            SIP tag: b1385811e50f0aai1
> |        Contact: <sip:asterisk at 19x.24x.16x.9x>
> |            Contact Binding: <sip:asterisk at 19x.24x.16x.9x>
> |                URI: <sip:asterisk at 19x.24x.16x.9x>
> |                    SIP contact address: sip:asterisk at 19x.24x.16x.9x
> |        Call-ID: 3ff0ae307575f1df61408b205af01196 at voip.convergenze.it
> |        CSeq: 102 ACK
> |        User-Agent: Convergenze VoGW1
> |        Max-Forwards: 16
> |        Content-Length: 0
> |
> |
> |and then
> |
> |Session Initiation Protocol
> |    Status-Line: SIP/2.0 400 Bad Request
> |        Status-Code: 400
> |        Resent Packet: False
> |    Message Header
> |        To: <sip:08281895109 at voip.convergenze.it>;tag=b1385811e50f0aai1
> |            SIP to address: sip:08281895109 at voip.convergenze.it
> |            SIP tag: b1385811e50f0aai1
> |        From: "anonymous" 
> <sip:asterisk at voip.convergenze.it>;tag=as6d07dd0a
> |            SIP Display info: "anonymous"
> |            SIP from address: sip:asterisk at voip.convergenze.it
> |            SIP tag: as6d07dd0a
> |        Call-ID: 3ff0ae307575f1df61408b205af01196 at voip.convergenze.it
> |        CSeq: 103 INVITE
> |        Via: SIP/2.0/UDP 19x.6x.19x.4x;branch=z9hG4bK9f79.7854d9e5.0
> |        Via: SIP/2.0/UDP 
> 19x.24x.16x.9x:5060;branch=z9hG4bK493c6be2;rport=5060
> |        Record-Route: <sip:19x.6x.19x.4x;ftag=as6d07dd0a;lr=on>
> |        Server: Sipura/SPA2100-3.2.5(d)
> |        Content-Length: 0
> |
> |
> |Any help is appreciated.
> |Regards
> |Rosairio
> |
> |
> |
> |
>
>