On Thursday 30 July 2009, ram wrote: > Hi > > I have setup rates in my table.. 0/0 for the profile 24hours basis > > and defined subscriber to use that profile to make rating for the outbound > calls. > > when the Opensips subscriber calls to PSTN Number 001732XXXXXX > > and wait for 2 or 3 rings and hangup the call. still i see the CDRtools > billing with rate.
The call was answered with a 200 OK, then ended with a BYE. Why exactly don't you expect to see it billed? > > > > *Signalling information* > > > <http://cdrtool.sbttalk.net/CDRTool/callsearch.phtml?cdr_source=opensips_radius&cdr_table=radius.radacct200907&order_by=RadAcctId&order_type=DESC&begin_datetime=1248904920&end_datetime=1248990900&maxrowsperpage=15&action=search&call_id=24271317073689-149641495610936%40202.63.111.2> > > Call id: > 24271317073689-149641495610...@x.x.x.2 > > > > From/to tags: > 2290420994/as2a1521b8 > > Start time: > 2009-07-30 02:06:55 > > Stop time: > 2009-07-30 02:07:09 > > Method: > Invite from ip-of-voipphone*:5060* > > From: > u...@domain.net > > Domain: > domain.net > > To (dialed URI): > 001732xxxx...@freeswitch.sbttalk.net > > Canonical URI: > 001732xxxx...@freeswitch.sbttalk.net > > Next hop URI: > 001732xxxx...@202.63.96.31 > > Destination: > USA (1732) > > Billing Party: > u...@domain.net > > Reseller: > 0 > > > > *Rating information* > > Duration: 14 s > App: audio > Destination: 1732 > Customer: subscriber=u...@domain.net > Connect: 0.0000 > StartTime: 2009-07-30 02:06:55 > -- > Span: 1 > Duration: 14 s > ProfileId: sl_standard / weekday > RateId: sl_standard / 0-24h > Rate: 0.0009 / 60 s > Price: 0.0002 > Price in: 0.0002 > -- > Price out: 0.0002 > Price in: 0.0002 > Margin: 0.0000 > > > > here is my siptrace > > > SIP trace on proxy cdrtool.domain for session > 24271317073689-149641495610...@voipphone-ip > -- > Packet 1 at from Opensip-IP to voipphone-ip (out) > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > voipphone-ip:5060;branch=z9hG4bK28385192501472111761;rport=5060 > From: user <sip:u...@domain.net:5060>;tag=2290420994 > To: 001732XXXXXX <sip:001732xxx...@domain.net:5060 > >;tag=c97b4d1cb1f3d0da549e06a8d482ef63.6b91 > Call-ID: 24271317073689-149641495610...@voipphone-ip > CSeq: 1 INVITE > Proxy-Authenticate: Digest realm="domain.net", > nonce="4a7162cd000001459588519a6132ccee82d5638acaecdff8" > Server: OpenSIPS (1.5.1-notls (i386/linux)) > Content-Length: 0 > Warning: 392 Opensip-IP:5060 "Noisy feedback tells: pid=17765 > req_src_ip=voipphone-ip req_src_port=5060 in_uri= > sip:001732xxx...@domain.net:5060 > out_uri=sip:001732xxx...@domain.net:5060via_cnt==1" > > --- > Packet 2 at from Opensip-IP to Opensip-IP (out) > INVITE sip:001732xxx...@opensip-ip:5062 SIP/2.0 > Record-Route: <sip:Opensip-IP;lr=on;did=ee6.8459b117> > Via: SIP/2.0/UDP Opensip-IP;branch=z9hG4bKe754.5dacff85.0 > Via: SIP/2.0/UDP > voipphone-ip:5060;received=voipphone- ip;branch=z9hG4bK938015010138926320;rport=5060 > From: user <sip:u...@domain.net:5060>;tag=2290420994 > To: 001732XXXXXX <sip:001732xxx...@domain.net:5060> > Call-ID: 24271317073689-149641495610...@voipphone-ip > CSeq: 2 INVITE > Contact: <sip:u...@voipphone-ip:5060> > Max-Forwards: 69 > Supported: replaces > User-Agent: Voip Phone 1.0 > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, > PRACK, UPDATE > Content-Type: application/sdp > Content-Length: 319 > P-hint: inbound->inbound > v=0 > o=4720779942 28362303 19011140 IN IP4 voipphone-ip > s=A conversation > c=IN IP4 voipphone-ip > t=0 0 > m=audio 10158 RTP/AVP 18 4 8 0 9 101 > a=rtpmap:18 G729/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > --- > Packet 3 at from Opensip-IP to Opensip-IP (in) > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > Opensip-IP;branch=z9hG4bKe754.5dacff85.0;received=Opensip-IP > Via: SIP/2.0/UDP > voipphone-ip:5060;received=voipphone- ip;branch=z9hG4bK938015010138926320;rport=5060 > Record-Route: <sip:Opensip-IP;lr=on;did=ee6.8459b117> > From: user <sip:u...@domain.net:5060>;tag=2290420994 > To: 001732XXXXXX <sip:001732xxx...@domain.net:5060> > Call-ID: 24271317073689-149641495610...@voipphone-ip > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:001732xxx...@opensip-ip:5062> > Content-Length: 0 > > --- > Packet 4 at from Opensip-IP to Opensip-IP (in) > SIP/2.0 200 OK > Via: SIP/2.0/UDP > Opensip-IP;branch=z9hG4bKe754.5dacff85.0;received=Opensip-IP > Via: SIP/2.0/UDP > voipphone-ip:5060;received=voipphone- ip;branch=z9hG4bK938015010138926320;rport=5060 > Record-Route: <sip:Opensip-IP;lr=on;did=ee6.8459b117> > From: user <sip:u...@domain.net:5060>;tag=2290420994 > To: 001732XXXXXX <sip:001732xxx...@domain.net:5060>;tag=as2a1521b8 > Call-ID: 24271317073689-149641495610...@voipphone-ip > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:001732xxx...@opensip-ip:5062> > Content-Type: application/sdp > Content-Length: 309 > v=0 > o=root 5836 5836 IN IP4 Opensip-IP > s=session > c=IN IP4 Opensip-IP > t=0 0 > m=audio 10004 RTP/AVP 18 0 8 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > --- > Packet 5 at from Opensip-IP to voipphone-ip (out) > SIP/2.0 200 OK > Via: SIP/2.0/UDP > voipphone-ip:5060;received=voipphone- ip;branch=z9hG4bK938015010138926320;rport=5060 > Record-Route: <sip:Opensip-IP;lr=on;did=ee6.8459b117> > From: user <sip:u...@domain.net:5060>;tag=2290420994 > To: 001732XXXXXX <sip:001732xxx...@domain.net:5060>;tag=as2a1521b8 > Call-ID: 24271317073689-149641495610...@voipphone-ip > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:001732xxx...@opensip-ip:5062> > Content-Type: application/sdp > Content-Length: 309 > v=0 > o=root 5836 5836 IN IP4 Opensip-IP > s=session > c=IN IP4 Opensip-IP > t=0 0 > m=audio 10004 RTP/AVP 18 0 8 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > --- > Packet 6 at from Opensip-IP to Opensip-IP (out) > BYE sip:001732xxx...@opensip-ip:5062 SIP/2.0 > Record-Route: <sip:Opensip-IP;lr=on> > Via: SIP/2.0/UDP Opensip-IP;branch=z9hG4bKf754.e5c10df7.0 > Via: SIP/2.0/UDP > voipphone-ip:5060;received=voipphone- ip;branch=z9hG4bK708313495288432690;rport=5060 > From: user <sip:u...@domain.net:5060>;tag=2290420994 > To: 001732XXXXXX <sip:001732xxx...@domain.net:5060>;tag=as2a1521b8 > Call-ID: 24271317073689-149641495610...@voipphone-ip > CSeq: 3 BYE > Max-Forwards: 69 > User-Agent: Voip Phone 1.0 > Content-Length: 0 > > --- > Packet 7 at from Opensip-IP to Opensip-IP (in) > SIP/2.0 200 OK > Via: SIP/2.0/UDP > Opensip-IP;branch=z9hG4bKf754.e5c10df7.0;received=Opensip-IP > Via: SIP/2.0/UDP > voipphone-ip:5060;received=voipphone- ip;branch=z9hG4bK708313495288432690;rport=5060 > Record-Route: <sip:Opensip-IP;lr=on> > From: user <sip:u...@domain.net:5060>;tag=2290420994 > To: 001732XXXXXX <sip:001732xxx...@domain.net:5060>;tag=as2a1521b8 > Call-ID: 24271317073689-149641495610...@voipphone-ip > CSeq: 3 BYE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:001732xxx...@opensip-ip:5062> > Content-Length: 0 > > --- > Packet 8 at from Opensip-IP to voipphone-ip (out) > SIP/2.0 200 OK > Via: SIP/2.0/UDP > voipphone-ip:5060;received=voipphone- ip;branch=z9hG4bK708313495288432690;rport=5060 > Record-Route: <sip:Opensip-IP;lr=on> > From: user <sip:u...@domain.net:5060>;tag=2290420994 > To: 001732XXXXXX <sip:001732xxx...@domain.net:5060>;tag=as2a1521b8 > Call-ID: 24271317073689-149641495610...@voipphone-ip > CSeq: 3 BYE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:001732xxx...@opensip-ip:5062> > Content-Length: 0 > > --- > -- Dan 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