[OpenSIPS-Users] CDRtools Billing missed calls..

ram talk2ram at gmail.com
Thu Jul 30 11:20:24 CEST 2009 

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.



     *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

---
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