[Serusers] BYE message to SER.
Marian Dumitru
marian.dumitru at voice-sistem.ro
Tue Oct 26 00:37:33 CEST 2004
Hi Ricardo,
I think there is something wrong about how the sipquest device does
routing based on RR and contact headers.
There are 2 interesting things in your dumps:
-In GW scenario, I don't see the Route inserted by SER (they have a
special format with ftag and lr params).
-In second scenario, there is no contact - both the ruri and Route
contain the address of ser. Probably because of this, SER cannot route
based on Route+contact and use usrloc which return 404.
Best regards,
Marian Dumitru
--
Voice Sistem
http://www.voice-sistem.ro
Ricardo Martinez wrote:
> Hello list.
> I have a problem-question regarding to the BYE message reaching SER.
> For the normail example i have a endpoint registered in SER and a
> PSTN-Gateway receiving the outgoing traffic. For this enviroment when i
> made a call from my endpoint to the PSTN gateway and the Gateway send the
> BYE (at the end of the call) this is what i see in the debug in SER.
>
> U gw1.mydomain.com:52154 -> sersip.mydomain.com:5060
> BYE sip:005622408196 at sersip.mydomain.com:5060 SIP/2.0
> Via: SIP/2.0/UDP gw1.mydomain.com:5060
> From: <sip:005622408196 at sersip.mydomain.com>;tag=A0272B49-B2D
> To: <sip:5555832351 at sersip.mydomain.com>;tag=ae4208f2a4
> Date: Mon, 25 Oct 2004 21:34:02 GMT
> Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9 at xx.xx.xx.42
> User-Agent: Cisco-SIPGateway/IOS-12.x
> Max-Forwards: 6
> Route: <sip:5555832351 at xx.xx.xx.42:5060>
> Timestamp: 1098740047
> CSeq: 101 BYE
> Content-Length: 0
>
>
> #
> U sersip.mydomain.com:5060 -> xx.xx.xx.42:5060
> BYE sip:5555832351 at xx.xx.xx.42:5060 SIP/2.0
> Record-Route: <sip:005622408196 at sersip.mydomain.com;ftag=A0272B49-B2D;lr=on>
> Via: SIP/2.0/UDP sersip.mydomain.com;branch=z9hG4bK4c65.37369754.0
> Via: SIP/2.0/UDP gw1.mydomain.com:5060
> From: <sip:005622408196 at 64.76.148.231>;tag=A0272B49-B2D
> To: <sip:5555832351 at sersip.mydomain.com>;tag=ae4208f2a4
> Date: Mon, 25 Oct 2004 21:34:02 GMT
> Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9 at xx.xx.xx.42
> User-Agent: Cisco-SIPGateway/IOS-12.x
> Max-Forwards: 5
> Timestamp: 1098740047
> CSeq:101 BYE
> Content-Length: 0
>
>
>
> #
> U xx.xx.xx.42:5060 -> sersip.mydomain.com:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP sersip.mydomain.com;branch=z9hG4bK4c65.37369754.0
> Via: SIP/2.0/UDP gw1.mydomain.com:5060
> From: <sip:005622408196 at sersip.mydomain.com>;tag=A0272B49-B2D
> To: <sip:5555832351 at sersip.mydomain.com>;tag=ae4208f2a4
> Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9 at xx.xx.xx.42
> CSeq: 101 BYE
> User-Agent: AddPac SIP Gateway
> Content-Length: 0
> Record-Route: <sip:005622408196 at sersip.mydomain.com;ftag=A0272B49-B2D;lr=on>
>
>
>
> #
> U sersip.mydomain.com:5060 -> gw1.mydomain.com:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP gw1.mydomain.com:5060
> From: <sip:005622408196 at sersip.mydomain.com>;tag=A0272B49-B2D
> To: <sip:5555832351 at sersip.mydomain.com>;tag=ae4208f2a4
> Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9 at xx.xx.xx.42
> CSeq: 101 BYE
> User-Agent: AddPac SIP Gateway
> Content-Length: 0
> Record-Route: <sip:005622408196 at sersip.mydomain.com;ftag=A0272B49-B2D;lr=on>
>
> As you can see the call is ended succesfully and there is no problem.
> Now, instead of having a PSTN-Gateway i use a SIP-H323 translator and all
> the calls are going out through this device. When i made a call from my
> endpoint and the SIP-H323 send the BYE message (at the end of the call) the
> debug shows this :
>
>
> U sipquest.mydomain.com:5060 -> sersip.mydomain.com:5060
> BYE sip:1112001003 at sersip.mydomain.com;ftag=25421eefa4;lr=on SIP/2.0
> Via: SIP/2.0/UDP sipquest.mydomain.com:5060;
> branch=z9hG4bKsKpiQfXRAwAYAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAB_
> From: <sip:1112001003 at sersip.mydomain.com>;tag=3896741524
> To: <sip:5555832351 at sersip.mydomain.com>;tag=25421eefa4
> CSeq: 1 BYE
> Call-ID: 25229c42-16c9-1e95-81ef-0002a400f1e9 at xx.xx.xx.42
> Contact: <sip:1112001003 at sipquest.mydomain.com>
> Route: <sip:5555832351 at sersip.mydomain.com>
> Max-Forwards: 69
> Content-Length: 0
>
>
> #
> U sersip.mydomain.com:5060 -> sipquest.mydomain.com:5060
> SIP/2.0 404 Not Found IT!
> Via: SIP/2.0/UDP sipquest.mydomain.com:5060;
> branch=z9hG4bKsKpiQfXRAwAYAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAB_;received=sipquest
> .mydomain.com
> From: <sip:1112001003 at sersip.mydomain.com>;tag=3896741524
> To: <sip:5555832351 at sersip.mydomain.com>;tag=25421eefa4
> CSeq: 1 BYE
> Call-ID: 25229c42-16c9-1e95-81ef-0002a400f1e9 at xx.xx.xx.42
> Server: Sip EXpress router (0.8.14 (i386/linux))
> Content-Length: 0
> Warning: 392 sersip.mydomain.com:5060 "Noisy feedback tells: pid=15285
> req_src_ip=sipquest.mydomain.com
> req_src_port=5060in_uri=sip:1112001003 at sersip.mydomain.com;ftag=25421eefa4;l
> r=on out_uri=sip:1112001003 at sersip.mydomain.com;ftag=25421eefa4;lr=on
> via_cnt==1"
>
>
>
> My question is : Why the call is ended succesfully in the first case and in
> the second case SER answer with a "NOT FOUND"? I only have registered in my
> location table the endpoints. Here is my ser.cfg file.
>
> Can someone please help me?.
>
>
>
> Ricardo Martinez
>
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