similar to: one-way audio

Hi, I've tried to make toll-free DID work for the last 2-3 weeks. Apparently only the IAX.CC/Sixtel personnel can make a call to my toll-free. Anybody else just gets a busy signal. It takes for them about 5-6 business days to respond to my request. It seems they are looking in the once a week to the High priority tickets. It also seems that they are permanently closed ! No matter when I

Hi, I want to assign different accountcodes (for billing) according to the IP address and or the H.323 name (chan_oh323). I tried in extensions.conf something like setVar(ACCOUNTCODE=userid) but in cdr I find the accountcode set in oh323.conf. Howto change it in extensions.conf? Roger.

Is it possible to modify CDR variables before insert into MySQL or CVS? how? thanks all; Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: alexm.vcf Type: text/x-vcard Size: 334 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060116/5a095cdc/alexm.vcf

Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1 # Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kan?le sle=$4 # Timeout bis zum n?chsten Versuch if [ -z $4 ]; then sle=0 fi s=1

Dear I am using a2billing accounting software, how can I charge on the destination target not at the caller side Ex: if user A have 10$ and user B have 10$ ,and the onnet call charge cost 1$ When user A call user B for 1 minute ,user A amount remains 10$ and user B amount be 9$ Regards ********************************************* No employee or agent is authorized to conclude

Hello- Is it possible to set the CDR record field called "accountcode" from within the dialplan? Or is there another way to cause this field to be set, preferably without using AGI code. Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England www.evtmedia.com

Dear all, I've searched the web about Asterisk with Radius integration for user authentication, and got a bit confused... I see that there have been some work around it, there is PortaOne's Radius client patch, an still open branch of Digium Issue Tracker "SIP peer authentication on an external database (RADIUS - LDAP)", etc. Although, none of these seems to give me the

We are using DISA with local SIP users. The user enters in a 2 digit code then they get a dialtone and the phone dials out. The problem is that the calls waits 10 seconds after the outgoing number is dialed, no matter what I put for the timeout values. Anyone else using DISA that has run into this? exten => _2X,1,Answer exten => _2X,2,DigitTimeout(2) exten =>

When I attempt to start asterisk with my modem setup listed it will not start attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated. running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org) just e-mail me privately if you need more info Thanks in

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

hello list, I subscribe to Sixtel.net for a DID just to see how it worked. they say DID active inmediatly , but after 4 days , I have no DID , I tried to call tech support office and it seems they all ways are in hollidays, a support ticket no good luck, I call back service also not good luck. so anyone here has experience with them? are they a SCAM? fortunally only put there $10 bucks but

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

My cdrs are missing accountcodes for incoming calls from other asterisk servers.. I've seen a few people mentioning this on the list and the solution seems to be setting up a dialplan for incoming calls from a particular sip peer.. in my opinion this does not scale well at all and I am looking for a solution to correct this problem. example sip peer: [asterisk_gw] type=friend

Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say "successfully", I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to

I am using a2billing as billing software ,and I make an 800 call service which means that the destination extension should be build I put this code at extensions.conf exten => 99909994,1,SetAccount(2704714849) exten => 99909994,2,Wait,2 exten => 99909994,3,DeadAGI(a2billingp.php) exten => 99909994,4,Wait,2 exten => 99909994,5,Hangup its not stable ,its works for 3 times

Greetings all, I'm not real good with dial plan programming and need some help. I've looked at the 2nd edition of the Asterisk book about GotoIf() and have a basic idea what I need to do but not sure about the correct way or the best way, to set it up. I need to branch based on whether the dialed number is long distance (international or not) or not. I have branch offices on SIP and IAX

I've setup cdr_mysql and am using AGI to authenticate users based on the called-from # (callerid), use the AGI perl module. Looking at the info stored in the caller detail, I see a field called "accountcode", is it possible for me to set this field in AGI? I'd like to tie it to a username, that I pull during my SQL authentication, so I can search the cdr table based on a

Hi folks, Sorry to post bunch of messages today, I am deep into this and hope any help from the experts. I am using the command Authenticate as explained in wi-ki: =================== Authenticate(password[|options]) Requires a user to enter a given password in order to continue execution. If the password begins with the '/' character, it is interpreted as a file which contains a list