[Serusers] Fwd: SIP 479 Regretfully

[flavio]

SER did not proxy INVITE to GW, so I did not see anything on the otherside.
However I've stopped SER, rebuild user via SERWEB and now all works fine.
Thanks a lot for support .



2007/5/31, olivier.taylor <olivier.taylor at gmail.com>:
>
>  you must have a way to debug gateway side, have a look there, sip trace
> doesn't give me any idea.
>
>  Olivier
>
>  flavio a écrit :
>  ---------- Forwarded message ----------
> From: flavio <flavio.patria at gmail.com>
> Date: 31-mag-2007 18.04
> Subject: Re: [Serusers] SIP 479 Regretfully
> To: olivier.taylor at hh174.be
>
>
> But If I try with my BT102 GrandStream (configured as Polycom on my
> SER) I'm able to start a call to PSTN Number.
> How is it possible? Have you any suggestions about?
>
> Thanks,
>
> ps I've also asterisk running on the same machine listening on port 5062.
>
> U 2007/05/31 18:00:52.457140 10.28.19.124:5060 -> 10.28.19.202:5060
> INVITE sip:0672020949 at 10.28.19.202 SIP/2.0.
> Via: SIP/2.0/UDP
> 10.28.19.124;branch=z9hG4bK5654e41ba6af4166.
> From: <sip:0660522016 at 10.28.19.202>;tag=44dcaa39db672de9.
> To: <sip:0672020949 at 10.28.19.202>.
> Contact: <sip:0660522016 at 10.28.19.124>.
> Supported: replaces.
> Proxy-Authorization: Digest username="0660522016",
> realm="10.28.19.202", algorithm=MD5,
> uri="sip:0672020949 at 10.28.19.202",
> nonce="465f0e8002ccdf57a20d382a49036bd0e5c691d0",
> response="554b818d1ecfc8455f1bc2e774881508".
> Call-ID: 19d87a9eb5a4d432 at 10.28.19.124.
> CSeq: 17072 INVITE.
> User-Agent: Grandstream BT110 1.0.8.12.
> Max-Forwards: 70.
> Allow:
> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
> Content-Type: application/sdp.
> Content-Length: 211.
> .
> v=0.
> o=0660522016 8000 8001 IN IP4 10.28.19.124.
> s=SIP Call.
> c=IN IP4 10.28.19.124.
> t=0 0.
> m=audio 5004 RTP/AVP 18 8 0.
> a=sendrecv.
> a=rtpmap:18 G729/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=ptime:20.
>
>
> 2007/5/31, olivier.taylor <olivier.taylor at gmail.com>:
>
>
>  hi,
>
> Regretfully, we were not able to process the URI
>
> probably a malformed URI, don't you have to remove the leading 0 and add the
> international code?
>
> hope it helps,
>
> Olivier
>
>
>
> flavio a écrit :
>
>
>  Hi to all.
> I've configured my polycom ip500 IPphone to use it with ser.
> If I try a call to users registred to ser all works fine.
> If I try to call PSTN Number through my gateway I've the follow sip message:
>
>
>
> INVITE sip:0672028405 at 10.28.19.202:5060;user=phone SIP/2.0.
> Via: SIP/2.0/UDP
> 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD.
> From: "0660522015" <sip:0660522015 at 10.28.19.202>;tag=8DDF0DB-E8BCEB84.
> To: <sip:0672028405 at 10.28.19.202;user=phone>.
> CSeq: 2 INVITE.
> Call-ID: 264aa927-d2a3a7c9-64f3fe3a at 10.28.19.143.
> Contact: <sip:0660522015 at 10.28.19.143>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER.
> User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1.
> Supported: 100rel,replace.
> Allow-Events: talk,hold,conference.
> Proxy-Authorization: Digest username="0660522015",
> realm="10.28.19.202",
> nonce="465ef44bbb09c0155dc8555314519a20cac896f8",
> uri="sip:0672028405 at 10.28.19.202:5060;user=phone",
> response="e6eed258e095ee6be5be6c92210f9d99", algorithm=MD5.
> Max-Forwards: 70.
> Content-Type: application/sdp.
> Content-Length: 237.
> .
> v=0.
> o=- 1180620549 1180620549 IN IP4 10.28.19.143.
> s=Polycom IP Phone.
> c=IN IP4 10.28.19.143.
> t=0 0.
> m=audio 2234 RTP/AVP 18 8 0 101.
> a=rtpmap:18 G729/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
>
> #
> U 2007/05/31 16:09:03.527834 10.28.19.202:5060 -> 10.28.19.143:5060
> SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL).
> Via: SIP/2.0/UDP
> 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD.
> From: "0660522015" <sip:0660522015 at 10.28.19.202>;tag=8DDF0DB-E8BCEB84.
> To:
> <sip:0672028405 at 10.28.19.202;user=phone>;tag=979d95a734c13f6db8b9e3a72b9f44a0.14e7.
> CSeq: 2 INVITE.
> Call-ID: 264aa927-d2a3a7c9-64f3fe3a at 10.28.19.143.
> Server: Sip EXpress router (0.9.6 (i386/linux)).
> Content-Length: 0.
> Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=8314
> req_src_ip=10.28.19.143 req_src_port=5060
> in_uri=sip:0672028405 at 10.28.19.202:5060;user=phone
> out_uri=sip:0672028405 at 10.28.52.105:5060:5060;user=phone
> via_cnt==1".
>
>
> Have you any suggestion about?
> I use my Polycom with Asterisk and BroadSoft without any problem.
>
> Thanks for your support and great patience :D
>
> Bye,
>
> F.
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>
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>
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* (o<     ing. Patria Flavio
* //\      phone 0823451358
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