<div>Hi Kaloyan,</div>
<div> </div>
<div>I think I understand what it means.</div>
<div>Thank you for your help. </div>
<div> </div>
<div>Best regards,</div>
<div>Charles<br><br> </div>
<div><span class="gmail_quote">2008/3/27, Kaloyan Kovachev <<a href="mailto:kkovachev@varna.net">kkovachev@varna.net</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">On Thu, 27 Mar 2008 21:19:01 +0800, Charles Wang wrote<br>> Hi, Kaloyan<br>> Can you please give me the last diff or tell me where to get it?<br>
<br>sure <a href="http://my.varna.net/dwl/rtcc-1.4.diff">http://my.varna.net/dwl/rtcc-1.4.diff</a><br><br>> Please also give me some tips to configure it.<br>> The <a href="http://asterisk-backports.org">asterisk-backports.org</a> seems be closed.<br>
> And can I ask some questions about the non-curl version?<br>> 1. Can I only set the results as channel variable instead of global variable?<br><br>yes, but please not the way variable is set (in multiple lines) in order to<br>
pass the variables for proccesing on execution<br><br>> 2. Is the unit of RTCC_INTERVAL 1/1000 second(ms) not 1 second ?<br><br>it is in milliseconds - preserving the original behavior of L() option<br><br>> 3. Assume that orginal the first parameter TimeLimit of L option is<br>
120000(120 seconds). Then, RTCC_APP should recheck the URL(test.php) after<br>${LIMIT_RECHECK_INTERVAL} milliseconds such as 6000 is 6 sec.<br>> Does it mean the TimeLimit will be added 10 sec if the result of<br>URL(test.php) is 10? And the TimeLimit is 130 sec.<br>
<br>yes and this will happen each 6 sec, so after 1 minute you will end with<br>220sec limit.<br>On each execution the number of seconds returned from test.php will be added<br>to the TimeLimit and the negative numbers will result in reducing the TimeLimit<br>
<br>> Best regards,<br>> Charles<br>><br>><br>> 2008/3/27, Kaloyan Kovachev <<a href="mailto:kkovachev@varna.net">kkovachev@varna.net</a>>: Hi,<br>> the curl version was provided from greyvoip, so probably he will be able to<br>
> tell you more about it.<br>> If you use some of the non curl versions (i may send you the latest diff if<br>> you wish), you may achieve the same results by setting in advance (as global<br>> variable):<br>><br>
> LIMIT_RECHECK_APP=$<br>> LIMIT_RECHECK_APP=${LIMIT_RECHECK_APP}{CURL(<a href="http://127.0.0.1/test.php">http://127.0.0.1/test.php</a>?<br>> LIMIT_RECHECK_APP=${LIMIT_RECHECK_APP}app=rtcc&accountcode=$<br>> LIMIT_RECHECK_APP=${LIMIT_RECHECK_APP}{ACCOUNTCODE}&dst=$<br>
> LIMIT_RECHECK_APP=${LIMIT_RECHECK_APP}{EXTEN}&channelid=$<br>> LIMIT_RECHECK_APP=${LIMIT_RECHECK_APP}{UNIQUEID}&seqnum=1)}<br>><br>> and you change your dial command to:<br>><br>> exten => _X.,n,Dial(SIP/1025,,L(${TimeLimit}:${RTCC_INTERVAL}))<br>
><br>> it should work, but keep in mind that it requires at least parameters to L()<br>> and that 60000 is 60sec not 6sec.<br>> As your script is returning fixed number (120 or 80 which for the curl<br>> version is in milliseconds, but in seconds for the non curl version) this will<br>
> cause your call duration to be increased with that amount of<br>> milliseconds/seconds each time. I would suggest for the tests to start with<br>> ${TimeLimit} set to 120000, ${RTCC_INTERVAL} being 10000 and your script<br>
> returning '-10'.<br>><br>> On Thu, 27 Mar 2008 18:10:40 +0800, Charles Wang wrote<br>> > Hi all,<br>> ><br>> > I tried to using rtcc-curl-1.4.13.patch in bug id 6335<br>> <a href="http://bugs.digium.com/view.php?id=6335">http://bugs.digium.com/view.php?id=6335</a> reported by KNK. I patch it to<br>
> asterisk <a href="http://1.4.18.1">1.4.18.1</a> and it seems work.<br>> ><br>> > My extensions.conf lists below:<br>> ><br>> > exten =><br>><br>_X.,1,Set(TimeLimit=${CURL(<a href="http://127.0.0.1/test.php?app=rtcc&accountcode=${ACCOUNTCODE}&dst=${EXTEN}&channelid=${UNIQUEID}&seqnum=1)}">http://127.0.0.1/test.php?app=rtcc&accountcode=${ACCOUNTCODE}&dst=${EXTEN}&channelid=${UNIQUEID}&seqnum=1)}</a>)<br>
<br>><br>> > exten => _X.,n,Set(TimeLimit=${MATH(${TimeLimit}+5,int)})<br>> > exten => _X.,n,Set(TimeLimit=${MATH(${TimeLimit}*1000,int)})<br>> > exten => _X.,n,Set(dst=${EXTEN})<br>> > exten => _X.,n,NoOp(Initial time limit for ${ACCOUNTCODE} and ${EXTEN} set<br>
> at ${TimeLimit})<br>> > exten => _X.,n,Set(RTCC_START_SEQNUM=2)<br>> > exten => _X.,n,Set(RTCC_INTERVAL=60000)<br>> > exten => _X.,n,Dial(SIP/1025,,L(${TimeLimit}:::<a href="http://127.0.0.1/test.php)">http://127.0.0.1/test.php)</a>)<br>
> > exten => _X.,n,Hangup<br>> ><br>> > My URL test.php always reponses interger 120. It is pure text format without<br>> any symbol before/after it.<br>> ><br>> > My test.php: ( one row only )<br>
> > 120<br>> ><br>> > -- Accepting AUTHENTICATED call from XXX.XXX.XXX.XXX:<br>> > > requested format = ilbc,<br>> > > requested prefs = (),<br>> > > actual format = ilbc,<br>
> > > host prefs = (ilbc),<br>> > > priority = mine<br>> > -- Executing [_X.@default:1] Set("SIP/2922-10", "TimeLimit=120") in new<br>> stack<br>> > -- Executing [_X.@default:2] Set("SIP/2922-10", "TimeLimit=125") in new<br>
> stack<br>> > -- Executing [_X.@default:3] Set("SIP/2922-10", "TimeLimit=125000") in<br>> new stack<br>> > -- Executing [_X.@default:4] Set("SIP/2922-10", "dst=295") in new stack<br>
> > -- Executing [_X.@default:5] NoOp("SIP/2922-10", "Initial time limit<br>> for and 295 set at 45000") in new stack<br>> > -- Executing [_X.@default:6] Set("SIP/2922-10", "RTCC_START_SEQNUM=2")<br>
> in new stack<br>> > -- Executing [_X.@default:7] Set("SIP/2922-10", "RTCC_INTERVAL=60000")<br>> in new stack<br>> > -- Executing [_X.@default:8] Dial("SIP/2922-10",<br>
> "SIP/1025||L(125000::<a href="http://127.0.0.1/test.php)">http://127.0.0.1/test.php)</a>") in new stack<br>> > -- Limit Data for this call:<br>> > > timelimit = 125000<br>> > > play_warning = 0<br>
> > > play_to_caller = yes<br>> > > play_to_callee = no<br>> > > warning_freq = 0<br>> > > rtcc url = //127.0.0.1/test.php<br>> > > rtcc interval = 60000<br>
> > > rtcc exp intvl = 0<br>> > > rtcc seqnum = 2<br>> > > start_sound = (null)<br>> > > warning_sound = timeleft<br>> > > end_sound = (null)<br>
> ><br>> > During the period, I trace the /var/log/httpd/access_log. I can't find any<br>> request to test.php. Should it be visited each 6 sec ?<br>> ><br>> > (Only this line)<br>> > <a href="http://127.0.0.1">127.0.0.1</a> - - [27/Mar/2008:17:51:54 +0800] "GET<br>
> /test.php?app=rtcc&accountcode=&dst=295&channelid=1206611514.2&seqnum=1<br>> HTTP/1.1" 200 3 "-" "asterisk-libcurl-agent/1.0"<br>> ><br>> > Then, I tried to reduce the integer number 120 to 80. I wish it can be<br>
> hunup when 80 seconds reached. But the answer was NO. It made my asterisk<br>> crashed. I got this message in debug mode.<br>> ><br>> > [Mar 27 17:52:58] DEBUG[32053]: app_dial.c:877 rtcccallback: call control<br>
> accountcode=2922, dst=295.<br>> > asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_dial.so:<br>> undefined symbol: curl_easy_init<br>> ><br>> > Can anyone kindly give me any idea?<br>
> ><br>> > Best regards,<br>> > Charles<br>> ><br>> > 2007/7/15, Grey Man <<a href="mailto:greyvoip@yahoo.com.au">greyvoip@yahoo.com.au</a>>:<br>> ><br>> > ----- Original Message ----<br>
> > From: Kaloyan Kovachev <<a href="mailto:kkovachev@varna.net">kkovachev@varna.net</a>><br>> > To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>
> > Sent: Sunday, 15 July, 2007 10:50:29 AM<br>> > Subject: Re: [asterisk-dev] Real-time call control for Dial app<br>> ><br>> > On Sat, 14 Jul 2007 14:36:27 -0700 (PDT), Grey Man wrote<br>> > > ----- Original Message ----<br>
> > > >From: Kaloyan Kovachev <<a href="mailto:kkovachev@varna.net">kkovachev@varna.net</a>><br>> > > >To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>
> > > >Sent: Saturday, 14 July, 2007 3:00:54 PM<br>> > > >Subject: Re: [asterisk-dev] Real-time call control for Dial app<br>> > > ><br>> > > >Hi again,<br>> > > >i guess Asterisk is not used for prepaid applications too much, or more<br>
> > > >likely the risk of overused account is just ignored (there are providers<br>> with<br>> > > >'pay as go' services with which i had negative balance for a while).<br>> > > >Even separate thread for each call bridge is not the best way and adding<br>
> > > >another one is not a good idea - agree, but the call control thread is<br>> > > >sleeping most of the time, so it shouldn't cause too much problems.<br>> > > >Unfortunately my C skills (and available time) are not enough to make (my<br>
> > > >long term idea) a single call control thread for which each call will just<br>> > > >register and which will also be responsible for the warning messages on<br>that<br>> > > >call instead of the bridging thread itself. I think this is the way to<br>
> go, but<br>> > > >for now this is at least some way to control the call duration after it<br>has<br>> > > >started.<br>> > ><br>> > > Hi Kaloyan,<br>> > ><br>> > > My idea is to put the real-time call control onto the the thread in<br>
> > channel.c that is already monitoring the bridge. You already nicely<br>slotted in<br>> > the "recheck" thread into this main bridge thread with your patch by using<br>the<br>> > nexteventts property and I think I might be able to do call control I need on<br>
> > the bridge thread and remove the need for the "recheck" thread.<br>> > ><br>> > > The problem with doing things on channel.c is it rightly doesn't know<br>> > anything about applications, dialplans or specific channel properties.<br>
However<br>> > the more I've played around with real-time call control the more I'm thinking<br>> > the main requirement is to have a clean scalable way to update the call time<br>> > and not so much about being able to periodically call more involved<br>
> > applications on an in progress calls. For example my requirement would be<br>> > satisifed by having the bridge thread on channel.c send the accountcode, call<br>> > destination and call time to an external IP socket and get back a single<br>
> > integer that specifies any adjustment that should be made to the call time.<br>> ><br>> > > Can't you do this via Manager? By leaving blank the LIMIT_RECHECK_APP, but<br>> > > setting LIMIT_RECHECK_INTERVAL and LIMIT_RECHECK_DELAY you will get Manager<br>
> > > event and then you may have enough time (LIMIT_RECHECK_DELAY) to proces it<br>> and<br>> > > to set CALL_LIMIT variable to that channel back from Manager. As you<br>need the<br>> > > acount code you will need to add it to the event field.<br>
> ><br>> > MAPI is an option but it would result in another moving part to be able to<br>> control calls and would still need the extra thread per bridged call to fire<br>> the events. If the real-time call control mechanism was a request approach<br>
> from channel.c then there is need to use MAPI or any additional threads. The<br>> bridge thread in channel.c doesn't seem to be doing much at all, just waiting<br>> for the call time limit to expire, so giving it an extra task would give<br>
> better utilisation of that thread.<br>> ><br>> > Regards,<br>> ><br>> > Greyman.<br>> ><br>> ><br>><br>____________________________________________________________________________________<br>
<br>> Yahoo!7 Mail has just got even bigger and better with unlimited storage on all<br>> webmail accounts.<br>> > <a href="http://au.docs.yahoo.com/mail/unlimitedstorage.html">http://au.docs.yahoo.com/mail/unlimitedstorage.html</a><br>
> ><br>> > _______________________________________________<br>> > --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br>> ><br>> > asterisk-dev mailing list<br>
> > To UNSUBSCRIBE or update options visit:<br>> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>> ><br>> ><br>> > --<br>
> ><br>> > Best Regards<br>> > Charles<br>><br>> _______________________________________________<br>> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br>
><br>> asterisk-dev mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>
><br>><br>> --<br>><br>> Best Regards<br>> Charles<br><br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br>
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<br><br clear="all"><br>-- <br><br>Best Regards<br>Charles