search for: settuping

Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really

Dears, i make a samba to DC in my enviroment test, i use howto manual samba in: https://wiki.samba.org/index.php/Setting_up_Samba_as_an_Active_Directory_Domain_Controller . many steps work, but when i created zone "./samba-tool dns zonecreate 172.16.0.2 0.99.10.in-addr.arpa" the stdout password is not: Password for [administrator at SAMDOM.EXAMPLE.COM]: The stdout is: Password for

Hi all, in my settup i decided to set mail_home in 10-mail.conf, and let dovecot do the hashing to a 2-level directory structure. mail_location = mdbox:~/mdbox:ALT=/altstorage/%h/mdbox mail_home = /vmail/%1Mu/%2.1Mu/%u In my userdb ist homedirectory not set and everythink works as expected, except when i use rawlog i get only logs in ~/dovecot.rawlog if home is set in userdb. In wiki

I've just tried setting up a Shrike (9) version of Redhat. Using the medium settings of lokkit, then adding manually accept commands for ports 137/udp 138/udp, 139/tcp and 445/tcp, I thought I should have been ready to go. This isn't the case, however. I know it's not the smb.conf settup because when I kill iptables samba works. When iptables IS running however, it will respond

please help settup font for run visual foxro 6 I from to vietnamese I use language vietnam But when settup wine I no fine .vntime regards, Mr. Qua

Hi all, We are planning to shift our sip users from one platform to another. (basically from one asterisk server to another). the problem we are facing is that both asterisk servers are using different ports to listen for sip. and both have live customers on them. provisioning their ata's is not a good option for us coz of our settup. we cant just ask the customers to change their ports for

We i have settup it like this it giveme caller id agent id and date-time on gsm file but i want them to be in folder on every day basis datewise. exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb) Any Idea ? Faisal > ------------------------------ > > Message: 16 >

I attempted to update an Avaya 4602 phone to the latest SIP firmware and now the phone stops at the bootloader. It keeps requesting an appsip.ebin file from my HTTP server and is no longer checking my TFTP server for update files. Since no appsip.ebin file was included in the firmware update provided by Avaya, I have no idea where I am supposed to get this file. And also since it is no

Hi, We're planning to launch a video stream. We do not have anything yet (no camera, no computer, no software, only a Linux server that will do the broadcasting with icecast installed). Has anyone ever done such a thing? What hardware do you suggest? What software can be used (both for Linux and Windows on source side). Can Icecast be used for this video streaming purpose (on server side)?

On Sun, Jul 24, 2005 at 10:19:51PM +0200, Balint Jacint wrote: > Hi, > > We're planning to launch a video stream. > We do not have anything yet (no camera, no computer, no software, only a > Linux server that will do the broadcasting with icecast installed). > Has anyone ever done such a thing? Yes! > What hardware do you suggest? What I've used several miniDV

The code is in utils/rawlog.c but it never seems to be called anywhere?

Hello Currently i'm using AIX 4.3.3. I've installed OpenSSH version 2.9p1 which by the way works with the way ive hacked it together to get X Displays working correctly. I can ssh into the system as myself export my display back no problem. I can ssh to a system as myself and ( su - userid ) and export my display back but i had to hack this together in order to get it to work. I

Hello guys, Software: Asterisk From CVS, Linux RH9 No telephony hardware. I have a SIP service provider (addaline.com) and (at this time) one Windows SIP (soft)phone. Problem: Both from console and from the sip phone if I dial an autside number it doesn't work. With sip debug I seen that addaline.com answer with a (401 Unauthorized) message when Asterisk issue a INVITE without

Hi guys ... I have a problem ... here is my settup luci xen0 xen1 [root at xen1 ~]# clustat Cluster Status for XEN_Cluster @ Mon Jun 29 03:31:23 2009 Member Status: Quorate Member Name ID Status ------ ---- ---- ------ xen0.genx.local

I have fitted a proportional odds model, but would like to compare it to a continuation ratio model. However, I am unable to fit the CR model _including_ imputated data. I guess my troubles start with settuping the data for the CR model. Any hint is appreciated! Christian library(Hmisc) library(rms) library(mice) ## simulating data (taken from rms::residuals.lrm) set.seed(1) n <- 400 age <- rnorm(n, 50, 10) blood.pressure <- rnorm(n, 120, 15) L <- .05*(age-50) + .03*(blood.pressure-120) p...

Listmates, Hopefully this is a compiz only question. Using the mouse wheel to scroll down web pages seems to be much slower in compiz than in kde. It also affects scrolling through long directories in konqueror (ex: /usr/share/doc/packages) but to a lesser extent than web content. Is there any setting, fix or workaround that might help? -- David C. Rankin, J.D., P.E. Rankin Law Firm, PLLC

Dear list, It has been a true adventure setting up a samba4 ad with a bind9 backend. >From what I can see, everything is more or less working: --> samba itself: root at bubba3-one:/etc/sssd# smbclient //localhost/netlogon -UAdministrator -c 'ls' Enter Administrator's password: Domain=[EARTH] OS=[Unix] Server=[Samba 4.1.4-SerNet-Debian-7.wheezy] .

Hello List! We have problems setting the right caller id on outgoing calls. The Asterisk Pbx is located in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the local telefon number 40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID numbers available. The telco is aspecting a 3 digit long Callerid from us, for example like "710", for the extension 10.

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", "SIP/1014|40|t") in new stack

Hello All I'm settup my asterisk as belows: sangoma card, connected with E1, CAS Signalling. I have two problem. 1. The asterisk don't received any DTMF when caller input to 2. when i dial to system, the caller hear bad sounds. monitor on console. asterisk show error. Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', but no exception