similar to: Re: astpp database creation failed...please help...

Hello, I'm setting up AST Post Paid application, is there anybody who set up astpp ? I followed the directions, i visited the astpp admin page in my web browser. But i couldn't setup the brands and routes etc. "Database unavailable -- please check configuration" appeared on the top of the page, so i went to "configure" section, I filled in the blanks according to my

Does anyone have experience setting up ASTPP? I have an Asterisk server in my office that I also give access to some friends and family that live outside Mexico so they can make local calls. I want to keep track of the costs and I only need to use ASTPP to rate the calls, not for calling cards or anything else. I found the documentation a little vague on the details so after setting up and

Hi, I have installed latest stable version of Asterisk. I registered 1 Quintum Gateway and 5 eyeBeam SoftPhone. But I can make just one call at the same time, if i try to make calls from 2 softphones to anotherone, second caller listens " the person have extension xxxx is on the phone ...." . So we couldn't make two or more calls at the same time for a SoftPhone. What should we

Is it possible that a RTP session between two end users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used "canreinvite=yes" but it didn't work. Description from asterisk conf. File; (canreinvite=yes ; allow RTP voice traffic to bypass Asterisk) Thanks Erdem HAKI - erdemh@tesas.com -------------- next part

Hello, I would like apply different call rate (tarif) per outgoing number (or group of phones, based on prefixes), I'm playing with astpp, but seems, that this feature isn't available here, can you recommend any other open-source billing (A2billing, AstBill?), that this can do? thank you! PJ

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

Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:.

I have a question. I have two numbers on Asterisk like 902121234567 and 902123645789 and i want to divert first number's call to Trunk if second number is unregistered. Is it possible? ?f yes, how? Flow Diagram: *Two numbers are registered on Asterisk 902121234567---------------------------- registered to Asterisk

Hi list, I set up two asterisk servers , 1001 is the first asterisk server's sip user, and 2001 is the second asterisk server's sip user. Each of them work well, but I don't konw how to connect them. I want to let the user in 1th Asterisk can call the user in 2nd Asterisk. First asterisk server ip : 192.168.3.101 Second asterisk server ip : 192.168.3.102 can someone

Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with "no, this is absolutely not for doing call marketing". I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students

Hi, What do I have to do to get local\number to work in a context? It works from my [from-internal]... however from subcontexts it does not work: Jan 6 15:55:32 VERBOSE[20237] logger.c: -- AGI Script Executing Application: (Dial) Options: (Local/570323xxxx) Jan 6 15:55:32 NOTICE[20237] chan_local.c: No such extension/context 570323xxxx@default creating local channel Jan 6 15:55:32

Good Day, I'm please to announce ASTPP (Asterisk Post Paid) billing version 1.2 Beta. I've spent the past months working on this software and have added several new features. Here is a partial list of new features: Partially redesigned web interface Automatic DID mapping Least Cost Routing Improved support for tying into webstores Screenshots available @

Does anyone have a mysql query that will compare a number from the asterisk cdr to a table of international country+city codes to determine the closest match? The two fields are; 1. Asterisk mysql cdr 'dst' field - sample record value '011441316551212' 2. rate table data like this DialPattern 011447977 011447979 011447980 011447981 011447984 011447985 011447986

I do part time consulting work. I need to setup an asterisk system to allow me to record both inbound and outbound calls to clients. I have one client that is just a PITA. The client has changed their mind three times so far and we are at step one. I have a spare slackware box and a seperate phone line for the consulting work. I have MCI Neighorhood as my carrier. What I need to know is: 1.

Hello Haki I fixed this problem following the instructions in /usr/src/zaptel-1.0.9/README.udev. Regards Cec?lia -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]Em nome de Erdem HAKI Enviada em: sexta-feira, 15 de julho de 2005 05:11 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Assunto:

Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote

I am copying the Master.csv file to another server and importing to mysql. I am looking for a simple billing application that will produce a bill for a give account code for a give period, based on a rate table. Is this available? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM:

Does anyone knows a working FOSS rating application for Asterisk? I tried CDRTool, but since it's free for non commercial use, it won't suit me. Also I gave a shot at Trabas (the one everybody says it sucks), but I couldn't understand how does it pull records from Asterisk's CDR table. The rate-engine addon (http://www.voip-info.org/tiki-index.php?page=Asterisk+addon+rate-engine)

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 all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? Thanks, Bryan Mahin Please visit us @