similar to: 1.6 and 1.8 version & A2Billing

Hey there, I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at Att, Flavio Roberto

Hello, We have a strange situation (asterisk 1.6.2.14), where we get a result for DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL. This is the (relevant) test dialplan: -------------------------------- [incoming-private] exten => _X., n, Dial(SIP/1001,30) exten => _X., n, NoOp(${DIALSTATUS}) exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1) [incoming-status] exten

Has any one used this? U(x[^arg[^...]]): x - Name of the subroutine to execute via Gosub arg - Arguments for the Gosub routine Execute via Gosub the routine <x> for the *called* channel before connecting to the calling channel. Arguments can be specified to the Gosub using '^' as a delimiter. The Gosub routine can set the variable ${GO

Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one

Hello all, I need that Asterisk Always use Reason in a CANCEL. How to do? thank you *Fran?ois * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100521/f3a91f36/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: francois.vcf Type: text/x-vcard Size: 400

Dear All. I want to become a wholesale VoIP traffic Provider , and i don't have a experience about the software used this career . I ask about Freeside billing system , FreeRADIUS AAA server and Asterisk telephony server gave me all i need to start my business . thanks -- Best Regards Mohamed Daif -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/xxxxxxxxxxxx ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound

Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These

Dear sir or madam, I am a 4-th year student at Yerevan State University, Armenia; and I am studying LLVM in order to write my Bachelor thesis. I am trying to write an llvm pass that just removes all "Add" commands and gives some statstics. Nevertheless, I get this segmentation fault: ................some rows about functions, that are not changed by my pass. The errors occurs after it

Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider) Now, with Asterisk2Billing would be something like this? exten => _2XXXXXXXX,1,Answer exten => _2XXXXXXXX,2,Wait,2 exten => _2XXXXXXXX,3,DeadAGI,a2billing.php exten => _2XXXXXXXX,4,Wait,2 exten =>

Hello all! I'm currently using Asterisk in conjunction with a2billing and everything seems to be working great so far. Now, all I'm missing is some sort of a GUI to monitor all calls going out through my trunks. I can always do 'sip show channels' or 'sip debug' from the console but I was wondering if there's anything that basically does the same thing but in a nicer,

Hi, i have the following data.frame : > Data[1:3,] dt amt geoTree merTree ref 1 0.71002484 3.334570 A2b B2b 0 2 0.49074936 2.544464 A2b B1a 0 3 0.06223433 3.617133 A1b B2a 0 i want to convert it to a list, like this: list(Data[1,],Data[2,],Data[3,]) [[1]] dt amt geoTree merTree ref 1 0.07333459 0.969585 A2a

Hi everyone, Please forgive me if my question is simple and my code terrible, I'm new to R. I am not looking for a ready-made answer, but I would really appreciate it if someone could share conceptual hints for programming, or point me toward an R function/package that could speed up my processing time. Thanks a lot for your help! ## My dataframe includes the variables 'year',

Hi All, new to the list. Wondering if anyone has / knows of, a good rate importer tool that can be used to standardize and normalize the ratesheets / rate decks etc. obtained from various carriers so they can be analysed and imported into a DB or be saved as a CSV or something? Thanks so much in advance aeg -------------- next part -------------- An HTML attachment was scrubbed... URL:

Suppose I have the following dataframe called test: test<-data.frame(year=rep(1990:2003,5),id=gl(5,length(1990:2003)),eif=as.vector(sapply(1:5,function(z){a<-rep(0,length(1990:2003));a[sample(1:length(1990:2003),sample(1:2,1))]<-1;a}))) year id eif 1990 1 0 1991 1 0 1992 1 0 2000 1 1 1994 1 0 1995 1 0 2001 1 0 1997 1 1 .... I want to create a new variable in

Hello R-users,    I have a data frame whose names of columns I don't know a priori, but the user of my code will know them. The user is supposed to save the name of the column that will need some reordering of the levels of the factor later on. The name of the column will be saved in an object called: variab the data frame is called df. If I try to the do following:

Hi Vardan I did same as you told and deleted the SIP information in Astdb and restarted asterisk. but the result was same. as you said there might be mistake in sip.conf so i am pasting both servers configuration here.. 1- nasir.server.com [abc] username=abc type=friend secret=mysecret nat=yes mailbox=12234568 incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=payasyougo

Hi All, I am newbie in this asterisk and a2billing technology . i had successfully installed asterisk in my server fedora -8 [server behind NAT/STUN] i after installation i can able to create users and tested the call features with X-Lite . the was working fine . after i installed the A2Billing in my same server with follow the steps from a2billing installation guide. but u cant access the

Hello; I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the

I am running into an issue with A2Billing. I will explain first of all that everything else works! the system is 90% complete its just this one small problem I am running into. So my problem is that when I place a call, 1. I dial my number that I want and A2Billing gets activated 2. it asks for my pin, upon successful entry of my pin A2Billing then 3. prompts me for my phone number then 4.