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: <[EMAIL PROTECTED]> To: <asterisk-users@lists.digium.com> Sent: Tuesday, September 09, 2008 10:30 PM Subject: asterisk-users Digest, Vol 50, Issue 22 > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. Re: Video on Hold? ([EMAIL PROTECTED]) > 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: [EMAIL PROTECTED] > Subject: Re: [asterisk-users] Video on Hold? > To: Asterisk Users Mailing List - Non-Commercial > Discussion<asterisk-users@lists.digium.com> > Message-ID: > <[EMAIL PROTECTED]> > > 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 > <[EMAIL PROTECTED]>: -------------- > > >> >> 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 >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20080909/c2617240/attachment-0001.htm > > ------------------------------ > > Message: 2 > Date: Tue, 9 Sep 2008 10:14:16 -0400 > From: "Dean Collins" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Asterisk and Network Monitoring > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <[EMAIL PROTECTED]> > 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Michiel > van Baak > Sent: Tuesday, 9 September 2008 9:29 AM > To: asterisk-users@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 > [EMAIL PROTECTED] > 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" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Asterisk and Network Monitoring > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > 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 > [EMAIL PROTECTED] > Bureau of Economic and Business Research > University of Florida > (352) 392-0171 Ext. 221 > > > >> -----Original Message----- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Michiel >> van Baak >> Sent: Tuesday, 9 September 2008 9:29 AM >> To: asterisk-users@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 >> [EMAIL PROTECTED] >> 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)" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Asterisk and Network Monitoring > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > 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: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Michiel >> van Baak >> Sent: Tuesday, 9 September 2008 9:29 AM >> To: asterisk-users@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 <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Asterisk and Network Monitoring > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="utf-8" > > http://www.voip-info.org/wiki/view/Asterisk+monitoring > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20080909/c1153c9e/attachment-0001.htm > > ------------------------------ > > Message: 6 > Date: Tue, 9 Sep 2008 10:34:31 -0400 > From: "Kristian Kielhofner" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Does X-Lite 'remember' Congestion state? > (halfway OT) > To: [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial > Discussion" <asterisk-users@lists.digium.com> > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > On Tue, Sep 9, 2008 at 9:42 AM, Paul Schewietzek <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Asterisk and Network Monitoring > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > 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 > [EMAIL PROTECTED] > 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 <[EMAIL PROTECTED]> > Subject: [asterisk-users] PRI auto-configure - continued from DEV list > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > On Tue, Sep 9, 2008 at 7:17 AM, Bill Michaelson <[EMAIL PROTECTED]> 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. > > > > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: smime.p7s > Type: application/x-pkcs7-signature > Size: 3234 bytes > Desc: S/MIME Cryptographic Signature > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20080909/214e11b1/attachment-0001.bin > > ------------------------------ > > Message: 9 > Date: Tue, 9 Sep 2008 10:30:16 -0500 > From: "Steven Sokol" <[EMAIL PROTECTED]> > Subject: [asterisk-users] AstriCon 2008 - Two Weeks To Go - Register > Today > To: "Asterisk Users" <asterisk-users@lists.digium.com> > Message-ID: > <[EMAIL PROTECTED]> > 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" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] OT: ARI > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]@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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> > Date: Mon, September 08, 2008 11:56 pm > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com>, [EMAIL PROTECTED] > > > 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 > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20080909/efd27002/attachment-0001.htm > > ------------------------------ > > Message: 11 > Date: Tue, 9 Sep 2008 22:00:37 +0530 > From: "Sriram" <[EMAIL PROTECTED]> > Subject: [asterisk-users] Asterisk - Operator switch billing > To: <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > 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 > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20080909/d6c5d6c5/attachment-0001.htm > > ------------------------------ > > Message: 12 > Date: Tue, 09 Sep 2008 12:47:41 -0400 > From: Josiah Bryan <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Asterisk - Operator switch billing > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > 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 >> >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> -- 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: 13 > Date: Tue, 9 Sep 2008 18:49:13 +0200 (CEST) > From: Julien Claassen <[EMAIL PROTECTED]> > Subject: [asterisk-users] CLI and AGI question > To: asterisk users mailinglist <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > 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 > > > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of asterisk-users Digest, Vol 50, Issue 22 > ********************************************** > _______________________________________________ -- 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