<div>Hi</div>
<div> </div>
<div>I have setup rates in my table.. 0/0 for the profile 24hours basis</div>
<div> </div>
<div>and defined subscriber to use that profile to make rating for the outbound calls.</div>
<div> </div>
<div>when the Opensips subscriber calls to PSTN Number 001732XXXXXX</div>
<div> </div>
<div>and wait for 2 or 3 rings and hangup the call. still i see the CDRtools billing with rate.</div>
<div> </div>
<div> </div>
<div> </div>
<div>
<table class="extrainfo" id="row1" style="DISPLAY: block" cellspacing="0" cellpadding="0" bgcolor="#ccddff" border="0">
<tbody>
<tr>
<td valign="top">
<table cellspacing="0" cellpadding="0" border="0">
<tbody>
<tr>
<td colspan="3">
<div><b>Signalling information</b></div></td></tr>
<tr>
<td width="10">
<div> </div></td>
<td colspan="2">
<div> </div><a href="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"><font color="orange"></font></a></td>
</tr>
<tr>
<td width="10">
<div> </div></td>
<td>
<div>Call id: </div></td>
<td>
<div><a href="mailto:24271317073689-149641495610936@x.x.x.2">24271317073689-149641495610936@x.x.x.2</a></div></td></tr>
<tr>
<td width="10">
<div> </div></td>
<td colspan="2">
<div> </div></td></tr>
<tr>
<td width="10">
<div> </div></td>
<td>
<div>From/to tags: </div></td>
<td>
<div>2290420994/as2a1521b8</div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>Start time: </div></td>
<td>
<div>2009-07-30 02:06:55 </div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>Stop time: </div></td>
<td>
<div>2009-07-30 02:07:09</div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>Method:</div></td>
<td>
<div>Invite from ip-of-voipphone<i>:5060</i> </div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>From:</div></td>
<td>
<div><a href="mailto:user@domain.net">user@domain.net</a></div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>Domain:</div></td>
<td>
<div><a href="http://domain.net">domain.net</a></div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>To (dialed URI):</div></td>
<td>
<div><a href="mailto:001732XXXXXXX@freeswitch.sbttalk.net">001732XXXXXXX@freeswitch.sbttalk.net</a></div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>Canonical URI: </div></td>
<td>
<div><a href="mailto:001732XXXXXXX@freeswitch.sbttalk.net">001732XXXXXXX@freeswitch.sbttalk.net</a></div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>Next hop URI:</div></td>
<td>
<div><a href="mailto:001732XXXXXXX@202.63.96.31">001732XXXXXXX@202.63.96.31</a></div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>Destination: </div></td>
<td>
<div>USA (1732)</div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>Billing Party:</div></td>
<td>
<div><font color="brown"><a href="mailto:user@domain.net">user@domain.net</a></font></div></td></tr>
<tr>
<td>
<div> </div></td>
<td>
<div>Reseller:</div></td>
<td>
<div><font color="brown">0</font></div></td></tr></tbody></table></td>
<td width="30">
<div> </div>
<td valign="top">
<div> </div></td>
<td width="30">
<div> </div></td>
<td valign="top">
<table cellspacing="0" cellpadding="0" border="0">
<tbody>
<tr>
<td colspan="3">
<div><b>Rating information</b></div></td></tr>
<tr>
<td>
<div> </div></td>
<td colspan="2">
<div>Duration: 14 s<br>App: audio<br>Destination: 1732<br>Customer: <a href="mailto:subscriber=user@domain.net">subscriber=user@domain.net</a><br>Connect: 0.0000<br>StartTime: 2009-07-30 02:06:55<br>--<br>Span: 1<br>Duration: 14 s<br>
ProfileId: sl_standard / weekday<br>RateId: sl_standard / 0-24h<br>Rate: 0.0009 / 60 s<br>Price: 0.0002<br>Price in: 0.0002<br>--<br>Price out: 0.0002<br>Price in: 0.0002<br>Margin: 0.0000</div></td></tr></tbody></table></td>
</td></tr></tbody></table></div>
<p> </p>
<div> </div>
<div>here is my siptrace </div>
<div> </div>
<div> </div>
<div>SIP trace on proxy cdrtool.domain for session <a href="mailto:24271317073689-149641495610936@voipphone-ip">24271317073689-149641495610936@voipphone-ip</a><br>--</div>
<div>Packet 1 at from Opensip-IP to voipphone-ip (out)</div>
<div>SIP/2.0 407 Proxy Authentication Required<br>Via: SIP/2.0/UDP voipphone-ip:5060;branch=z9hG4bK28385192501472111761;rport=5060<br>From: user <<a href="http://sip:user@domain.net:5060">sip:user@domain.net:5060</a>>;tag=2290420994<br>
To: 001732XXXXXX <<a href="http://sip:001732XXXXXX@domain.net:5060">sip:001732XXXXXX@domain.net:5060</a>>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.6b91<br>Call-ID: <a href="mailto:24271317073689-149641495610936@voipphone-ip">24271317073689-149641495610936@voipphone-ip</a><br>
CSeq: 1 INVITE<br>Proxy-Authenticate: Digest realm="<a href="http://domain.net">domain.net</a>", nonce="4a7162cd000001459588519a6132ccee82d5638acaecdff8"<br>Server: OpenSIPS (1.5.1-notls (i386/linux))<br>
Content-Length: 0<br>Warning: 392 Opensip-IP:5060 "Noisy feedback tells: pid=17765 req_src_ip=voipphone-ip req_src_port=5060 in_uri=<a href="http://sip:001732XXXXXX@domain.net:5060">sip:001732XXXXXX@domain.net:5060</a> out_uri=<a href="http://sip:001732XXXXXX@domain.net:5060">sip:001732XXXXXX@domain.net:5060</a> via_cnt==1"</div>
<div><br>---<br>Packet 2 at from Opensip-IP to Opensip-IP (out)</div>
<div>INVITE sip:001732XXXXXX@Opensip-IP:5062 SIP/2.0<br>Record-Route: <sip:Opensip-IP;lr=on;did=ee6.8459b117><br>Via: SIP/2.0/UDP Opensip-IP;branch=z9hG4bKe754.5dacff85.0<br>Via: SIP/2.0/UDP voipphone-ip:5060;received=voipphone-ip;branch=z9hG4bK938015010138926320;rport=5060<br>
From: user <<a href="http://sip:user@domain.net:5060">sip:user@domain.net:5060</a>>;tag=2290420994<br>To: 001732XXXXXX <<a href="http://sip:001732XXXXXX@domain.net:5060">sip:001732XXXXXX@domain.net:5060</a>><br>
Call-ID: <a href="mailto:24271317073689-149641495610936@voipphone-ip">24271317073689-149641495610936@voipphone-ip</a><br>CSeq: 2 INVITE<br>Contact: <sip:user@voipphone-ip:5060><br>Max-Forwards: 69<br>Supported: replaces<br>
User-Agent: Voip Phone 1.0<br>Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE<br>Content-Type: application/sdp<br>Content-Length: 319<br>P-hint: inbound->inbound </div>
<div>v=0<br>o=4720779942 28362303 19011140 IN IP4 voipphone-ip<br>s=A conversation<br>c=IN IP4 voipphone-ip<br>t=0 0<br>m=audio 10158 RTP/AVP 18 4 8 0 9 101<br>a=rtpmap:18 G729/8000<br>a=rtpmap:4 G723/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:0 PCMU/8000<br>a=rtpmap:9 G722/16000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=sendrecv</div>
<div>---<br>Packet 3 at from Opensip-IP to Opensip-IP (in)</div>
<div>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP Opensip-IP;branch=z9hG4bKe754.5dacff85.0;received=Opensip-IP<br>Via: SIP/2.0/UDP voipphone-ip:5060;received=voipphone-ip;branch=z9hG4bK938015010138926320;rport=5060<br>Record-Route: <sip:Opensip-IP;lr=on;did=ee6.8459b117><br>
From: user <<a href="http://sip:user@domain.net:5060">sip:user@domain.net:5060</a>>;tag=2290420994<br>To: 001732XXXXXX <<a href="http://sip:001732XXXXXX@domain.net:5060">sip:001732XXXXXX@domain.net:5060</a>><br>
Call-ID: <a href="mailto:24271317073689-149641495610936@voipphone-ip">24271317073689-149641495610936@voipphone-ip</a><br>CSeq: 2 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Contact: <sip:001732XXXXXX@Opensip-IP:5062><br>Content-Length: 0</div>
<div><br>---<br>Packet 4 at from Opensip-IP to Opensip-IP (in)</div>
<div>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP Opensip-IP;branch=z9hG4bKe754.5dacff85.0;received=Opensip-IP<br>Via: SIP/2.0/UDP voipphone-ip:5060;received=voipphone-ip;branch=z9hG4bK938015010138926320;rport=5060<br>Record-Route: <sip:Opensip-IP;lr=on;did=ee6.8459b117><br>
From: user <<a href="http://sip:user@domain.net:5060">sip:user@domain.net:5060</a>>;tag=2290420994<br>To: 001732XXXXXX <<a href="http://sip:001732XXXXXX@domain.net:5060">sip:001732XXXXXX@domain.net:5060</a>>;tag=as2a1521b8<br>
Call-ID: <a href="mailto:24271317073689-149641495610936@voipphone-ip">24271317073689-149641495610936@voipphone-ip</a><br>CSeq: 2 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Contact: <sip:001732XXXXXX@Opensip-IP:5062><br>Content-Type: application/sdp<br>Content-Length: 309</div>
<div>v=0<br>o=root 5836 5836 IN IP4 Opensip-IP<br>s=session<br>c=IN IP4 Opensip-IP<br>t=0 0<br>m=audio 10004 RTP/AVP 18 0 8 101<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv</div>
<div>---<br>Packet 5 at from Opensip-IP to voipphone-ip (out)</div>
<div>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP voipphone-ip:5060;received=voipphone-ip;branch=z9hG4bK938015010138926320;rport=5060<br>Record-Route: <sip:Opensip-IP;lr=on;did=ee6.8459b117><br>From: user <<a href="http://sip:user@domain.net:5060">sip:user@domain.net:5060</a>>;tag=2290420994<br>
To: 001732XXXXXX <<a href="http://sip:001732XXXXXX@domain.net:5060">sip:001732XXXXXX@domain.net:5060</a>>;tag=as2a1521b8<br>Call-ID: <a href="mailto:24271317073689-149641495610936@voipphone-ip">24271317073689-149641495610936@voipphone-ip</a><br>
CSeq: 2 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:001732XXXXXX@Opensip-IP:5062><br>Content-Type: application/sdp<br>
Content-Length: 309</div>
<div>v=0<br>o=root 5836 5836 IN IP4 Opensip-IP<br>s=session<br>c=IN IP4 Opensip-IP<br>t=0 0<br>m=audio 10004 RTP/AVP 18 0 8 101<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv</div>
<div>---<br>Packet 6 at from Opensip-IP to Opensip-IP (out)</div>
<div>BYE sip:001732XXXXXX@Opensip-IP:5062 SIP/2.0<br>Record-Route: <sip:Opensip-IP;lr=on><br>Via: SIP/2.0/UDP Opensip-IP;branch=z9hG4bKf754.e5c10df7.0<br>Via: SIP/2.0/UDP voipphone-ip:5060;received=voipphone-ip;branch=z9hG4bK708313495288432690;rport=5060<br>
From: user <<a href="http://sip:user@domain.net:5060">sip:user@domain.net:5060</a>>;tag=2290420994<br>To: 001732XXXXXX <<a href="http://sip:001732XXXXXX@domain.net:5060">sip:001732XXXXXX@domain.net:5060</a>>;tag=as2a1521b8<br>
Call-ID: <a href="mailto:24271317073689-149641495610936@voipphone-ip">24271317073689-149641495610936@voipphone-ip</a><br>CSeq: 3 BYE<br>Max-Forwards: 69<br>User-Agent: Voip Phone 1.0<br>Content-Length: 0</div>
<div><br>---<br>Packet 7 at from Opensip-IP to Opensip-IP (in)</div>
<div>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP Opensip-IP;branch=z9hG4bKf754.e5c10df7.0;received=Opensip-IP<br>Via: SIP/2.0/UDP voipphone-ip:5060;received=voipphone-ip;branch=z9hG4bK708313495288432690;rport=5060<br>Record-Route: <sip:Opensip-IP;lr=on><br>
From: user <<a href="http://sip:user@domain.net:5060">sip:user@domain.net:5060</a>>;tag=2290420994<br>To: 001732XXXXXX <<a href="http://sip:001732XXXXXX@domain.net:5060">sip:001732XXXXXX@domain.net:5060</a>>;tag=as2a1521b8<br>
Call-ID: <a href="mailto:24271317073689-149641495610936@voipphone-ip">24271317073689-149641495610936@voipphone-ip</a><br>CSeq: 3 BYE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Contact: <sip:001732XXXXXX@Opensip-IP:5062><br>Content-Length: 0</div>
<div><br>---<br>Packet 8 at from Opensip-IP to voipphone-ip (out)</div>
<div>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP voipphone-ip:5060;received=voipphone-ip;branch=z9hG4bK708313495288432690;rport=5060<br>Record-Route: <sip:Opensip-IP;lr=on><br>From: user <<a href="http://sip:user@domain.net:5060">sip:user@domain.net:5060</a>>;tag=2290420994<br>
To: 001732XXXXXX <<a href="http://sip:001732XXXXXX@domain.net:5060">sip:001732XXXXXX@domain.net:5060</a>>;tag=as2a1521b8<br>Call-ID: <a href="mailto:24271317073689-149641495610936@voipphone-ip">24271317073689-149641495610936@voipphone-ip</a><br>
CSeq: 3 BYE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:001732XXXXXX@Opensip-IP:5062><br>Content-Length: 0</div>
<div><br>---</div>
<div> </div>