[OpenSIPS-Users] how to implement call forward after 486 (busy) with 181 message

[Bogdan-Andrei Iancu]

So, I guess this 482 is generated by this 192.168.1.2 
<mailto:sip%3A13 at 192.168.1.2> right ? by looking at the message as it 
is, there is no reason for 192.168.1.2 <mailto:sip%3A13 at 192.168.1.2> to 
"see" a loop at all - a request can simply go several time through the 
same server (spiralling). A loop is defined like : a server may report a 
loop is the same request already visited it, coming from the same 
address and having the same RURI (before and now).

Regards,
Bogdan

Daniel Ribeiro wrote:
> sorry,
>
> INVITE sip:13 at 192.168.1.2 <mailto:sip%3A13 at 192.168.1.2> SIP/2.0
> Record-Route: <sip:192.168.1.200;lr=on>
> Record-Route: <sip:192.168.1.200;lr=on>
> Date: Fri, 19 Mar 2010 11:28:25 GMT
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 192.168.1.200;branch=z9hG4bK10b4.3a60a363.0
> Via: SIP/2.0/UDP 192.168.1.200;branch=z9hG4bK10b4.2a60a363.1
> Via: SIP/2.0/UDP 
> 10.1.8.16:5064;received=10.1.8.16;branch=z9hG4bK5e28c133-b831-df11-91ee-002421899f1c;rport=5064
> User-Agent: Ekiga/2.0.12
> From: "101816" <sip:116 at 192.168.1.200 
> <mailto:sip%3A116 at 192.168.1.200>>;tag=3c55c033-b831-df11-91ee-002421899f1c
> Call-ID: 0852c033-b831-df11-91ee-002421899f1c at danielribeiro
> To: <sip:11 at 192.168.1.200 <mailto:sip%3A11 at 192.168.1.200>>
> Contact: <sip:116 at 10.1.8.16:5062;transport=udp>
> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
> Content-Type: application/sdp
> Content-Length: 271
> Max-Forwards: 68
>
> v=0
> o=- 1268998105 1268998105 IN IP4 10.1.8.16
> s=Opal SIP Session
> c=IN IP4 10.1.8.16
> t=0 0
> m=audio 5020 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> m=video 5022 RTP/AVP 31
> a=rtpmap:31 H261/90000
>
>
> On Tue, Mar 23, 2010 at 8:55 AM, Bogdan-Andrei Iancu 
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>
>     Hi Daniel,
>
>     OK, but you haven't answered to my question - what is the first
>     line of
>     the request (that part is missing in the INVITE you posted).
>
>     Regards,
>     Bogdan
>
>     Daniel Ribeiro wrote:
>     > Hi Bogdan,
>     >
>     > The end of calls is an application based on SIP api running two
>     UA at
>     > the same instance at 192.168.1.2 : 5060.
>     > Thanks,
>     >
>     > Daniel
>     >
>     >
>     >
>     ------------------------------------------------------------------------
>     >
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>
>
>     --
>     Bogdan-Andrei Iancu
>     www.voice-system.ro <http://www.voice-system.ro>
>
>
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>
>
>
> -- 
> Daniel Ribeiro
> ------------------------------------------------------------------------
>
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-- 
Bogdan-Andrei Iancu
www.voice-system.ro