[OpenSIPS-Users] CDRtools Billing missed calls..

Dan Pascu dan at ag-projects.com
Fri Jul 31 08:13:31 CEST 2009 

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:
>  24271317073689-149641495610936 at 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:
>  user at domain.net
> 
>  Domain:
>  domain.net
> 
>  To (dialed URI):
>  001732XXXXXXX at freeswitch.sbttalk.net
> 
>  Canonical URI:
>  001732XXXXXXX at freeswitch.sbttalk.net
> 
>  Next hop URI:
>  001732XXXXXXX at 202.63.96.31
> 
>  Destination:
>  USA (1732)
> 
>  Billing Party:
>  user at domain.net
> 
>  Reseller:
>  0
> 
> 
> 
>    *Rating information*
> 
>  Duration: 14 s
> App: audio
> Destination: 1732
> Customer: subscriber=user at 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
> 24271317073689-149641495610936 at 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:user at domain.net:5060>;tag=2290420994
> To: 001732XXXXXX <sip:001732XXXXXX at domain.net:5060
> >;tag=c97b4d1cb1f3d0da549e06a8d482ef63.6b91
> Call-ID: 24271317073689-149641495610936 at 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:001732XXXXXX at domain.net:5060
> out_uri=sip:001732XXXXXX at domain.net:5060via_cnt==1"
> 
> ---
> Packet 2 at  from Opensip-IP to Opensip-IP (out)
> INVITE sip:001732XXXXXX at 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:user at domain.net:5060>;tag=2290420994
> To: 001732XXXXXX <sip:001732XXXXXX at domain.net:5060>
> Call-ID: 24271317073689-149641495610936 at voipphone-ip
> CSeq: 2 INVITE
> Contact: <sip:user at 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:user at domain.net:5060>;tag=2290420994
> To: 001732XXXXXX <sip:001732XXXXXX at domain.net:5060>
> Call-ID: 24271317073689-149641495610936 at voipphone-ip
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:001732XXXXXX at 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:user at domain.net:5060>;tag=2290420994
> To: 001732XXXXXX <sip:001732XXXXXX at domain.net:5060>;tag=as2a1521b8
> Call-ID: 24271317073689-149641495610936 at voipphone-ip
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:001732XXXXXX at 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:user at domain.net:5060>;tag=2290420994
> To: 001732XXXXXX <sip:001732XXXXXX at domain.net:5060>;tag=as2a1521b8
> Call-ID: 24271317073689-149641495610936 at voipphone-ip
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:001732XXXXXX at 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:001732XXXXXX at 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:user at domain.net:5060>;tag=2290420994
> To: 001732XXXXXX <sip:001732XXXXXX at domain.net:5060>;tag=as2a1521b8
> Call-ID: 24271317073689-149641495610936 at 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:user at domain.net:5060>;tag=2290420994
> To: 001732XXXXXX <sip:001732XXXXXX at domain.net:5060>;tag=as2a1521b8
> Call-ID: 24271317073689-149641495610936 at voipphone-ip
> CSeq: 3 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:001732XXXXXX at 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:user at domain.net:5060>;tag=2290420994
> To: 001732XXXXXX <sip:001732XXXXXX at domain.net:5060>;tag=as2a1521b8
> Call-ID: 24271317073689-149641495610936 at voipphone-ip
> CSeq: 3 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:001732XXXXXX at Opensip-IP:5062>
> Content-Length: 0
> 
> ---
> 


-- 
Dan

More information about the Users mailing list