[Serusers] Fwd: SIP 479 Regretfully

flavio flavio.patria at gmail.com
Thu May 31 18:06:21 CEST 2007


---------- 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.
> > _______________________________________________
> > Serusers mailing list
> > Serusers at lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
>


--
********************************
* (o<     ing. Patria Flavio
* //\      phone 0823451358
* V_/_  mobile 3407873357
*
********************************


-- 
********************************
* (o<     ing. Patria Flavio
* //\      phone 0823451358
* V_/_  mobile 3407873357
*
********************************


More information about the sr-users mailing list