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:
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|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:
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|Session Initiation Protocol
| Request-Line: ACK sip:0681140017@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@voip.convergenze.it>;tag=as6d07dd0a
| SIP Display info: "anonymous"
| SIP from address: sip:asterisk@voip.convergenze.it
| SIP tag: as6d07dd0a
| To: <sip:08281895109@voip.convergenze.it>;tag=b1385811e50f0aai1
| SIP to address: sip:08281895109@voip.convergenze.it
| SIP tag: b1385811e50f0aai1
| Contact: <sip:asterisk@19x.24x.16x.9x>
| Contact Binding: <sip:asterisk@19x.24x.16x.9x>
| URI: <sip:asterisk@19x.24x.16x.9x>
| SIP contact address: sip:asterisk@19x.24x.16x.9x
| Call-ID: 3ff0ae307575f1df61408b205af01196(a)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@voip.convergenze.it>;tag=b1385811e50f0aai1
| SIP to address: sip:08281895109@voip.convergenze.it
| SIP tag: b1385811e50f0aai1
| From: "anonymous"
<sip:asterisk@voip.convergenze.it>;tag=as6d07dd0a
| SIP Display info: "anonymous"
| SIP from address: sip:asterisk@voip.convergenze.it
| SIP tag: as6d07dd0a
| Call-ID: 3ff0ae307575f1df61408b205af01196(a)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
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