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:
[email protected]
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:
[email protected]
Domain:
domain.net
To (dialed URI):
[email protected]
Canonical URI:
[email protected]
Next hop URI:
[email protected]
Destination:
USA (1732)
Billing Party:
[email protected]
Reseller:
0
*Rating information*
Duration: 14 s
App: audio
Destination: 1732
Customer: [email protected]
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:[email protected]:5060>;tag=2290420994
To: 001732XXXXXX <sip:[email protected]: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:[email protected]:5060
out_uri=sip:[email protected]: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:[email protected]:5060>;tag=2290420994
To: 001732XXXXXX <sip:[email protected]: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:[email protected]:5060>;tag=2290420994
To: 001732XXXXXX <sip:[email protected]: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:[email protected]:5060>;tag=2290420994
To: 001732XXXXXX <sip:[email protected]: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:[email protected]:5060>;tag=2290420994
To: 001732XXXXXX <sip:[email protected]: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:[email protected]:5060>;tag=2290420994
To: 001732XXXXXX <sip:[email protected]: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:[email protected]:5060>;tag=2290420994
To: 001732XXXXXX <sip:[email protected]: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:[email protected]:5060>;tag=2290420994
To: 001732XXXXXX <sip:[email protected]: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
---
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