similar to: Billing Applications - When does the bill start??

Hi, Has anyone connected * to a Colt E1 line in Europe? If so could you send me the zaptel.conf and zapata.conf. Thanks, Aaron __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

Hi, Does anyone know anything about the following? In a hosted environment where several area DIDs are provisioned on a single server, how do most carriers establish the origination DID, number. Asterisk allows us to modify the CallerID, name, number and DNID channel variables before dialling out via SIP. Most carriers will allow us to spoof a callerID when placing a call, and pass it forward.

Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Our telecom provider (your communications) gives us monthly itemized bills that list

Hello ; how can i enable billing only while using specific trunk (ex:zap) but internal sip calls will not be counted specifically how to make all outbound is counted i am using asterisk mysql cdr enabled -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050626/0faf0974/attachment.htm

Hi to everybody, i tried to find an asnwer before posting this, but most astcc billing issues i searched refer to the case when no billing occurs at all. In my case i get only initial charges and any subsequent minute does not count for billing. In my iax.conf i entered the "notransfer = yes" but nothing changed. My test card and test calls are the following TEST-CARD en N/A

I have ASTCC installed, and compare it with NuFone, however, I find that the billing of NuFone is always a few secondes more (6 to 24 seconds) Does anybody has an explanation / solution for it? bye Ronald

Hi all, I have about 10 SIP phones for different users defined in sip.conf, each with their own accountcode= entry. There is a global setting in sip.conf that states amaflags=documentation There are 3 IAX->PSTN gateways defined in iax.conf for outbound calls. These do not have an accountcode=, but do have amaflags=billing defined in each. The theory was that all calls should be logged, those

Hello I have a samba 4.0.3 pdc and a samba 3.5.10 as a fileserver and i am having an issue that i like to share with you. I have a share son the samba 3 setup like this [Comercial] browsable = Yes comment = Comercial path = /shares2/Comercial valid users = @Ingenieria, @Mercadeo, @Comercial, @SIIF, @Costos, administrador, backup write list = @Comercial,

Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. True CDR,

Thank you very much, Max, for this valuable and informative answer. Offline billing must be quite complex to set up as several telco may be involved (or origination,transit or termination). Moving to normal landline fare seems much simpler ! Thanks again 2017-03-28 21:41 GMT+02:00 Max Grobecker <max.grobecker at ml.grobecker.info>: > Hi, > > in Germany, this kind of regulation

In United Kingdom, we have time based dialling pricing from most of Telco's based on time the call is placed! It is called PEAK (08.00- 18.00 Mon-Fri), OFF PEAK(18.00-08.00 Mon-Fri) and WEEKEND (all other times! Could someone from any of other countries let me know if time based charging exists in your country? Also, what numbers (up to 999) are commonly used for emergency, police or other

Hi, I suspected that I the analogue phone should have got a pass through signal when the power was off to the server, unfortunately it doesn't. I kept asking digium support about that but they didn't give me an answer. The problem is how do I identify whether the X100P is incompatibel with the network or faulty without possibly wasting another USD100??? Aaron On Sat, 2004-05-15, Eric

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

Ahhh this could be my problem! I just checked which wires on the RJ11 cable had a voltage across them and it was the yellow and green (3 & 4?). From what someone posted the other day it's supposed to be Bumble Bee and Christmas Tree. I did have to get a technician out to fix my line when it was first installed because it was dead. Maybe he wired it up incorrectly or maybe they just do it

Hi, Does anyone know of a list of internationally accessible PSTN talking clocks? I find talkjing clocks a good way to test the call quality to a particular country. There are a quite a few available in the US but the only other two countries I have found numbers for are the UK and Sweden. Other countries obviously have them but they generally don't seem accessible from international

Before I start, I just want to say this is not necessarily a problem with ASTCC, but may be a problem the way I have set up ASTCC (and possibly the way others have set it up as well). The issue is that ASTCC tries to match the pattern *anywhere* in the called number, not necessarily only at the beginning. I have set up ASTCC Routes like this: 1800 Tollfree Trunk1 0 0 100 1416 Canada Trunk2 0 0

Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

Good Day, I'm finally getting around to officially announcing ASTPP. For the last 6 months, I've been working on converting ASTCC into a decent billing package for asterisk. I'm just finishing up fixing a few bugs before the 1.0 release and would appreciate if there would be a few who would be willing to do some testing on the software. Here is a list of features: Provide call

Hola, ¿como están? Me encuentro haciendo un trabajo de Explotación de Información para la facultad y tengo un inconveniente a la hora de preparar los datos para poder operar con ellos. Tengo un atributo "Titulo Secundario" que cuenta con 139 valores diferentes (al final les dejo algunos de estos valores con la cantidad de ocurrencias de cada uno) y lo que quiero hacer es

Hello, My situation is that , I can't make calls with asterisk, but with x-lite works fine. Asterisk shows , that successfully registers with another SIP server, asterisk sends invite, gets trying, and after 30 secs asterisk gets 408 Request timeout. And as I said , with x-lite no problems. I heard that for comercial purposes, this SIP server detects asterisk , and ignores him. Or maybe it