[asterisk-users] Asterisk - Operator switch billing
Sriram
d_r_sriram at hotmail.com
Tue Sep 9 12:25:14 CDT 2008
Hi
I think i wasnt clear here - It'll be either a premium rate line/toll free
line but the customer should be charged Rs.6/- per minute only when he hears
a prompt(where it'll ask him to press 1 to continue) once he presses 1 to
accept the terms . Till the time he hears only the prompt the asterisk box
should not send the "reversal" to the billing switch.. only after pressing 1
should the charging begin...I hope am clear now
Any ways to implement this ?
Rgds
Sriram
----- Original Message -----
From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, September 09, 2008 10:30 PM
Subject: asterisk-users Digest, Vol 50, Issue 22
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> Today's Topics:
>
> 1. Re: Video on Hold? (k4rjj at bellsouth.net)
> 2. Re: Asterisk and Network Monitoring (Dean Collins)
> 3. Re: Asterisk and Network Monitoring (Martin Smith)
> 4. Re: Asterisk and Network Monitoring (Darrick Hartman (lists))
> 5. Re: Asterisk and Network Monitoring (EdPimentl)
> 6. Re: Does X-Lite 'remember' Congestion state? (halfway OT)
> (Kristian Kielhofner)
> 7. Re: Asterisk and Network Monitoring (Jay R. Ashworth)
> 8. PRI auto-configure - continued from DEV list (Bill Michaelson)
> 9. AstriCon 2008 - Two Weeks To Go - Register Today (Steven Sokol)
> 10. Re: OT: ARI (Mark Hamilton)
> 11. Asterisk - Operator switch billing (Sriram)
> 12. Re: Asterisk - Operator switch billing (Josiah Bryan)
> 13. CLI and AGI question (Julien Claassen)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 09 Sep 2008 14:11:50 +0000
> From: k4rjj at bellsouth.net
> Subject: Re: [asterisk-users] Video on Hold?
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion<asterisk-users at lists.digium.com>
> Message-ID:
> <090920081411.2495.48C6842600088CCD000009BF22230706129B0A02D2089B9A019C04040A0DBF06069DCB05 at att.net>
>
> Content-Type: text/plain; charset="us-ascii"
>
>
> Is the idea to switch to another video source or stay with the callers
> camera? An option for both would be nice. I could see a help desk
> placing a caller in que and a 1-2 min video coming on showing some simple
> video of "how to hook it up".
> -------------- Original message from Russell Bryant
> <russell at digium.com>: --------------
>
>
>>
>> On Sep 8, 2008, at 7:31 PM, Russell Bryant wrote:
>>
>> >
>> > On Sep 8, 2008, at 9:15 AM, Gordon Henderson wrote:
>> >> Does/Will asterisk support video streaming on hold?
>> >>
>> >> Been playing with videphones as of late, and a client asked about
>> >> video on
>> >> hold - standard MoH works fine - but on the target video phone the
>> >> image
>> >> just freezes - any way to inject a video?
>> >
>> >
>> > This is not something that is supported right now. However, it would
>> > be relatively straight forward to add for a developer interested in
>> > adding it.
>>
>> I just went and wrote a first draft in
>> http://svn.digium.com/svn/asterisk/team/russell/video_on_hold/
>> . I haven't tested it, yet, though. However, as soon as I can get
>> this tested and any issues fixed, it will be merged into Asterisk 1.6.
>>
>> This would be a fun project to finish up in the code zone at
>> Astricon. :)
>>
>> --
>> Russell Bryant
>> Senior Software Engineer
>> Open Source Team Lead
>> Digium, Inc.
>>
>>
>>
>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
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> ------------------------------
>
> Message: 2
> Date: Tue, 9 Sep 2008 10:14:16 -0400
> From: "Dean Collins" <Dean at cognation.net>
> Subject: Re: [asterisk-users] Asterisk and Network Monitoring
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <3685A8FD247FA94C957C4304AB386A040C063A at cognationsvr1.Cognation.local>
> Content-Type: text/plain; charset="US-ASCII"
>
> Has anyone ever 'released' an Asterisk module that is easily
> shared/downloadable?
>
> Or doesn't the nagios open source code work like that?
>
>
> Cheers,
>
> Dean
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michiel
> van Baak
> Sent: Tuesday, 9 September 2008 9:29 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Asterisk and Network Monitoring
>
> On 14:50, Tue 09 Sep 08, Jacobus van Niekerk wrote:
>> Dear Asterisk Users
>>
>> I'm looking for a solution that can be used to monitor Asterisk and
> the
>> Telco lines aswell as the network (Servers, WAN & LAN links, Router &
>> Switches)
>
> We use nagios for that.
>
> --
>
> Michiel van Baak
> michiel at vanbaak.eu
> http://michiel.vanbaak.eu
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
>
> "Why is it drug addicts and computer aficionados are both called users?"
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 9 Sep 2008 10:19:17 -0400
> From: "Martin Smith" <martins at bebr.ufl.edu>
> Subject: Re: [asterisk-users] Asterisk and Network Monitoring
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <2709A069CB844242A469ECC57C29D621028EF9B9 at smtp.ufl.edu>
> Content-Type: text/plain; charset="us-ascii"
>
> Nagios has a plugins and a plugins-extra/contrib section and I've seen
> *lots* of Asterisk plugins/checkers. As always, consult the Google and
> find it -- http://www.google.com/search?q=check_asterisk -- and
> Voip-Info also has a page on Nagios check scripts for Asterisk at
> http://www.voip-info.org/tiki-index.php?page=check_asterisk.
>
> Cheers all,
>
>
> Martin Smith, Systems Developer
> martins at bebr.ufl.edu
> Bureau of Economic and Business Research
> University of Florida
> (352) 392-0171 Ext. 221
>
>
>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
>> Dean Collins
>> Sent: Tuesday, September 09, 2008 10:14 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Asterisk and Network Monitoring
>>
>> Has anyone ever 'released' an Asterisk module that is easily
>> shared/downloadable?
>>
>> Or doesn't the nagios open source code work like that?
>>
>>
>> Cheers,
>>
>> Dean
>>
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michiel
>> van Baak
>> Sent: Tuesday, 9 September 2008 9:29 AM
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] Asterisk and Network Monitoring
>>
>> On 14:50, Tue 09 Sep 08, Jacobus van Niekerk wrote:
>> > Dear Asterisk Users
>> >
>> > I'm looking for a solution that can be used to monitor Asterisk and
>> the
>> > Telco lines aswell as the network (Servers, WAN & LAN
>> links, Router &
>> > Switches)
>>
>> We use nagios for that.
>>
>> --
>>
>> Michiel van Baak
>> michiel at vanbaak.eu
>> http://michiel.vanbaak.eu
>> GnuPG key:
>> http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
>>
>> "Why is it drug addicts and computer aficionados are both
>> called users?"
>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 09 Sep 2008 09:21:50 -0500
> From: "Darrick Hartman (lists)" <dhartman at djhsolutions.com>
> Subject: Re: [asterisk-users] Asterisk and Network Monitoring
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <48C6867E.2030704 at djhsolutions.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Dean,
>
> I'm using Zabbix to monitor network interfaces, storage, cpu load and a
> few other things on several asterisk boxes. I'm just looking at adding
> Asterisk specific monitoring. Simple things like sip registration is
> pretty easy. Getting the actual status of zap-daddy hardware might be a
> little trickier. When I get something together I can pass it along.
>
> Darrick
>
> Dean Collins wrote:
>> Has anyone ever 'released' an Asterisk module that is easily
>> shared/downloadable?
>>
>> Or doesn't the nagios open source code work like that?
>>
>>
>> Cheers,
>>
>> Dean
>>
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michiel
>> van Baak
>> Sent: Tuesday, 9 September 2008 9:29 AM
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] Asterisk and Network Monitoring
>>
>> On 14:50, Tue 09 Sep 08, Jacobus van Niekerk wrote:
>>> Dear Asterisk Users
>>>
>>> I'm looking for a solution that can be used to monitor Asterisk and
>> the
>>> Telco lines aswell as the network (Servers, WAN & LAN links, Router &
>>> Switches)
>>
>> We use nagios for that.
>>
>
> --
> Darrick Hartman
> DJH Solutions, LLC
> http://www.djhsolutions.com
> http://www.djhsolutions.com/wiki
>
>
>
> ------------------------------
>
> Message: 5
> Date: Tue, 9 Sep 2008 10:30:38 -0400
> From: EdPimentl <edpimentl at gmail.com>
> Subject: Re: [asterisk-users] Asterisk and Network Monitoring
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <9dc4a1670809090730m1fd13521rd42901a621f771c4 at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> http://www.voip-info.org/wiki/view/Asterisk+monitoring
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> ------------------------------
>
> Message: 6
> Date: Tue, 9 Sep 2008 10:34:31 -0400
> From: "Kristian Kielhofner" <kkielhofner at star2star.com>
> Subject: Re: [asterisk-users] Does X-Lite 'remember' Congestion state?
> (halfway OT)
> To: tetsuo2k6 at web.de, "Asterisk Users Mailing List - Non-Commercial
> Discussion" <asterisk-users at lists.digium.com>
> Message-ID:
> <2d9149cd0809090734g4213babdye6f82b6665f5d614 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On Tue, Sep 9, 2008 at 9:42 AM, Paul Schewietzek <tetsuo2k6 at web.de> wrote:
>> Hi all,
>>
>>
>>
>> I noticed a strange X-Lite behavior, it's connected to an asterisk box.
>> The client registers normally and everything works fine. When I dial out
>> (via E1-PRI) and the called party is unavailable, and asterisk indicates
>> CONGESTION to X-Lite. So far so good.
>>
>> When I try to make another call directly after that (doesn't matter if
>> the same or a different extension is being dialed), X-Lite again tells
>> me about unavailability, but on the asterisk console nothing happens, it
>> seems like X-Lite didn't even try to pass the call to asterisk. The only
>> way to immediately make another call is to restart X-Lite :(
>>
>> After waiting a few minutes, everything works fine again. This behavior
>> is reproducable.
>>
>> I wonder if X-Lite tries to 'remember' about the unavailability, because
>> it thinks 'Hey, we didn't get a connection two minutes ago, chances are
>> we won't get one now', which of course would be stupid when we dial a
>> different extension.
>>
>> Could it be that? Or do you think maybe I'm looking in the wrong
>> direction? Any ideas how to get around that behavior (X-Lite, as far as
>> I can see, has no options available regarding that issue)? Maybe
>> asterisk is able to say 'Don't think you're smart!' to the client phone
>> via SIP? (I don't know much about SIP internals)
>>
>>
>>
>> Kindest regards, Paul
>>
>
> We've been experiencing this behavior with CounterPath for a while
> now. They've acknowledged the bug but haven't provided a fix yet...
>
> If you restart the phone or wait about 10 minutes (I think) it should
> be able to make outbound calls again.
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
>
>
>
> ------------------------------
>
> Message: 7
> Date: Tue, 9 Sep 2008 10:45:59 -0400
> From: "Jay R. Ashworth" <jra at baylink.com>
> Subject: Re: [asterisk-users] Asterisk and Network Monitoring
> To: asterisk-users at lists.digium.com
> Message-ID: <20080909144559.GD23322 at cgi.jachomes.com>
> Content-Type: text/plain; charset=us-ascii
>
> On Tue, Sep 09, 2008 at 09:21:50AM -0500, Darrick Hartman (lists) wrote:
>> I'm using Zabbix to monitor network interfaces, storage, cpu load and a
>> few other things on several asterisk boxes. I'm just looking at adding
>> Asterisk specific monitoring. Simple things like sip registration is
>> pretty easy. Getting the actual status of zap-daddy hardware might be a
>> little trickier. When I get something together I can pass it along.
>
> $ head -1q /proc/zaptel/*
>
> Cheers,
> -- jra
> --
> Jay R. Ashworth Baylink
> jra at baylink.com
> Designer The Things I Think RFC
> 2100
> Ashworth & Associates http://baylink.pitas.com '87
> e24
> St Petersburg FL USA http://photo.imageinc.us +1 727 647
> 1274
>
> Those who cast the vote decide nothing.
> Those who count the vote decide everything.
> -- (Josef Stalin)
>
>
>
> ------------------------------
>
> Message: 8
> Date: Tue, 09 Sep 2008 11:13:45 -0400
> From: Bill Michaelson <bill at cosi.com>
> Subject: [asterisk-users] PRI auto-configure - continued from DEV list
> To: asterisk-users at lists.digium.com
> Message-ID: <48C692A9.6000105 at cosi.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Tue, Sep 9, 2008 at 7:17 AM, Bill Michaelson <bill at cosi.com> wrote:
>
>> > I'm faced with an installation at a client site with supposed PRI
>> > service on
>> > a fractional T1.
> Steve Totaro wrote:
>
> I usually configure the entire span of 24 channels (23 B + 1 D) and
> only the turned up channels go into service. This is good for a
> couple of reasons.
>
> 1. No configuration changes are needed if the client decides to
> "light up" some more B channels
> 2. All B channels that are lit up will come up but not the B chans
> that are not in service, so configuring the entire span in Asterisk
> will not effect anything negatively. Channels that do not come up are
> not used by Asterisk.
>
> I have had issues with this only once, the entire span came up, not
> just what was provisioned, so calls going out on those channels did
> not work. The carrier put a Cisco box at the demarc that was
> configured for a full PRI going to the Asterisk box.
>
> -------------------------
> Steve,
>
> Thanks, I like this idea, and I appreciate the tip. I will try it.
> Meanwhile, I'm finding from others' comments that it is extremely common
> to find the D channel on 24, which is primarily what concerned me - and my
> inability to divine this precisely in my case led to my suggestion/inquiry
> on the dev list. I've seen enough docs that indicate that the D channel
> could be anywhere in the group, also implying that it's not unlikely to be
> at 13 or 6, IIRC. I have visions of sitting in a lonely room repeatedly
> editing zaptel/zapata.conf and smacking it again, and again...
>
> Of course, due to my inability to assure everything else in the
> configuration is correct, I could do all that smacking for nothing. I want
> to eliminate variables or otherwise devise a logical step-by-step
> procedure for getting this running.
>
> In my case, I've got an Adtran TSU120e doing a split between the old
> Nortel PBX (which I'm trying to replace) and a Cisco router for the IP
> side of the service. From fiddling around with the Adtran panel, I've
> been able to determine that there are 12 channels being sent to the DSX-1,
> but it tells me no more than that. If I could safely assume that D is on
> 24, and configuring the other 23 per your suggestion will be OK, maybe
> there is hope.
>
>
>
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> ------------------------------
>
> Message: 9
> Date: Tue, 9 Sep 2008 10:30:16 -0500
> From: "Steven Sokol" <ssokol at sokol-associates.com>
> Subject: [asterisk-users] AstriCon 2008 - Two Weeks To Go - Register
> Today
> To: "Asterisk Users" <asterisk-users at lists.digium.com>
> Message-ID:
> <eebdd9aa0809090830k75e486d1oeebcaa46dd275d6 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Asterisk Users -
>
> Just a reminder that we're now only two weeks away from the kick-off
> of AstriCon 2008. This year's show is looking really good: at this
> point we're expecting over 700 Asterisk users, developers, and
> resellers. There are currently 60 presentations scheduled, including
> keynotes from Brian Aker from MySQL and Stefan ?berg from Skype. The
> exhibit floor has expanded with more than 30 vendors. The CodeZone
> (the development lab and lounge) will have more gear than ever before
> -- Digium is providing a huge array of servers, desktops and cards,
> plus many of the exhibitors have agreed to sacrifice hardware and
> software to the cause.
>
> If you've never been to an AstriCon before, ask some of those who
> have: it's a great opportunity to meet other Asterisk users, recruit
> talent, expand your sales channel and have a great time. The content
> is 100% on-target coverage of Asterisk or related industry
> events/trends (with far less "sales pitch" than most conferences).
> The exhibit hall vendors are all offering goods and services that are
> relevant to the Asterisk users, developers or resellers.
>
> Register here:
> http://www.astricon.net/2008/glendale/web/attendRegister.php
>
> If you've not yet registered, please get signed up as soon as
> possible. The main hotel is rapidly running out of rooms --
> fortunately there are plenty of alternate hotels within walking
> distance. Remember that prices go up by $100 once the conference has
> started.
>
> Thanks,
>
> -S
>
> --
> Steven Sokol
> Product Manager - Software Products
> Digium
>
> P.S. - To those on the Dev and Biz lists: sorry for the duplicate
> posting, but we really want everyone to know about the event.
>
>
>
> ------------------------------
>
> Message: 10
> Date: Tue, 9 Sep 2008 11:43:15 -0400
> From: "Mark Hamilton" <mark.h at cage151.com>
> Subject: Re: [asterisk-users] OT: ARI
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <007a01c91292$c6efcfa0$54cf6ee0$@h at cage151.com>
> Content-Type: text/plain; charset="utf-8"
>
> Steve,
>
>
>
> Thank you for that link!!
>
> However, you saying that it might not work scares me already.. :S
>
>
>
> I guess I?ll have to somehow try it out. It would be nice where a the
> install needs a block of code pasted into extensions.conf, and a block
> placed in /var/www/ and we?re good to go, lol. But now I be dreaming.
>
>
>
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mark
> Hamilton
> Sent: September 9, 2008 5:23 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] OT: ARI
>
>
>
> Paul,
>
> Thank you very much for your reply!
> Recordings and voicemail are not even the most important thing really, but
> call forwarding is. ARI seemed to have all of them mungled in, so I
> mentioned it.
>
> However, if you know of something that will require me to add a few
> contexts to the dialplan and put a webgui of sorts in, that would be
> really nice.
>
> I know you said this is not much help, but trust me.. it is. It's in the
> right direction atleast.
>
>
>
>
> -------- Original Message --------
> Subject: Re: [asterisk-users] OT: ARI
> From: Paul Hales <pdhales at optusnet.com.au>
> Date: Mon, September 08, 2008 11:56 pm
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>, mark.h at cage151.com
>
>
> ARI really only let people check their voicemail via a web interface -
> for CDR's you can install areske cdr interface as that bolts on to vanilla
> asterisk with a small amount of work.
>
> Recordings - how complicated an interface do you need? From memory
> there's something in the contribs folder that can help you out.
>
> With regards to call forwarding - that's a bit more tricky. You need an
> interface than can affect your dialplan.
>
> I know that's not a lot of help, but I'm trying to break it down into
> chunks, and then cross off the easy chunks first (or the hard ones,
> depending on your preference and the priority of the chunks)
>
> later,
>
> PaulH
>
>
>
> Mark Hamilton wrote:
>>
>> Hi,
>>
>>
>>
>> I?m looking for a GUI like ARI by LittleJohn Consulting (which is not
>> being maintained actively anymore, but FreePBX seems to include it) so
>> users can login, check cdrs, recordings, call forward, etc.
>>
>>
>>
>> Does anyone know of any such working app that can be integrated into
>> vanilla asterisk?
>>
>>
>>
>> Thanks,
>>
>> Mark
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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> ------------------------------
>
> Message: 11
> Date: Tue, 9 Sep 2008 22:00:37 +0530
> From: "Sriram" <d_r_sriram at hotmail.com>
> Subject: [asterisk-users] Asterisk - Operator switch billing
> To: <asterisk-users at lists.digium.com>
> Message-ID: <BLU114-DAV12F40AF508139421EDAEC7CF540 at phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi All
>
> I am a premium IVR content service provider thats runs on premium rate
> lines, my setup (currently on PRIs) is like customer dials the short code
> (premium number) which gets forwarded on the PRIs to my IVR. In the
> normal world the customer is charged immediately the call is answered by
> the IVR. On operator's new requirement - he wants me to design the IVR in
> such a way that a customer will call on a number (Toll Free/Premium Rate)
> but the billing will start only if his MSISDN is present on a database
> that he will give it to me...Is this sort of differential charging on a
> single call possible in Asterisk ? If yes how and what additional
> parameters do i need to get from him
>
> Please assist
>
> Thanks
> Sriram
>
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> ------------------------------
>
> Message: 12
> Date: Tue, 09 Sep 2008 12:47:41 -0400
> From: Josiah Bryan <jbryan at productiveconcepts.com>
> Subject: Re: [asterisk-users] Asterisk - Operator switch billing
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <48C6A8AD.7030005 at productiveconcepts.com>
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>
> A simple AGI script would be able to handle that easily, I would think.
> Or am I missing something in the details?
>
> -josiah
>
> Sriram wrote:
>> Hi All
>>
>> I am a premium IVR content service provider thats runs on premium rate
>> lines, my setup (currently on PRIs) is like customer dials the short
>> code (premium number) which gets forwarded on the PRIs to my IVR. In
>> the normal world the customer is charged immediately the call is
>> answered by the IVR. On operator's new requirement - he wants me to
>> design the IVR in such a way that a customer will call on a number (Toll
>> Free/Premium Rate) but the billing will start only if his MSISDN is
>> present on a database that he will give it to me...Is this sort of
>> differential charging on a single call possible in Asterisk ? If yes how
>> and what additional parameters do i need to get from him
>>
>> Please assist
>>
>> Thanks
>> Sriram
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
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>
>
>
>
> ------------------------------
>
> Message: 13
> Date: Tue, 9 Sep 2008 18:49:13 +0200 (CEST)
> From: Julien Claassen <julien at c-lab.de>
> Subject: [asterisk-users] CLI and AGI question
> To: asterisk users mailinglist <asterisk-users at lists.digium.com>
> Message-ID: <Pine.LNX.4.64L.0809091844570.12296 at arima>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
> Hello!
> I wondered could I (mis)use an AGI program to decide if I pickup. At the
> moment asterisk has to pck up, when the "ring tone" has stopped playing.
> The dialplan looks like this:
> *** CUT ***
> exten => NUM,1,System(mplayer file > /dev/null)
> exten => NUM,n,Answer()
> exten => NUM,n,Jack(i(system:playback_1)o(system:capture_1))
> *** CUT ***
> Now I wonder could I insert some AGI-script that would let me decide to
> pick
> up in my own time?
> Has anyone done this before/ I'm aonly going for it, because on my pc I
> use
> asterisk and nothig else and this seems to be the most comfortable
> solution in
> SO MANY respects.
> Kindest regards and thanks for anything
> Julien
>
> --------
> Music was my first love and it will be my last (John Miles)
>
> ======== FIND MY WEB-PROJECT AT: ========
> http://ltsb.sourceforge.net
> the Linux TextBased Studio guide
> ======= AND MY PERSONAL PAGES AT: =======
> http://www.juliencoder.de
>
>
>
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