[asterisk-dev] Real-time call control for Dial app

Charles Wang lazy.charles at gmail.com
Thu Mar 27 08:19:01 CDT 2008


Hi, Kaloyan

Can you please give me the last diff or tell me where to get it?
Please also give me some tips to configure it.
The asterisk-backports.org seems be closed.

And can I ask some questions about the non-curl version?

1. Can I only set the results as channel variable instead of global
variable?
2. Is the unit of RTCC_INTERVAL 1/1000 second(ms) not 1 second ?
3. Assume that orginal the first parameter TimeLimit of L option is
120000(120 seconds). Then, RTCC_APP should recheck the URL(test.php) after
${LIMIT_RECHECK_INTERVAL} milliseconds such as 6000 is 6 sec.
   Does it mean the TimeLimit will be added 10 sec if the result of URL(
test.php) is 10? And the TimeLimit is 130 sec.

Best regards,
Charles



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

Best Regards
Charles
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