search for: settup

...eparate 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 want to install another software to make this or modify all my settup. I'm wonder if someone is using something simple to limmit calls. Anyway if someone is using some other programs/software/scripts and another settup/method please let me know how is yours. I want to check few methods to realize that limmit. Thank you for help guys.

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...

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 to a connect request but not seem to accept the ID and password supplied. What am I missing?? North Lilly

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

...ur 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 registration (many of them dont know what a port is) Is it possible to make single asterisk server listen on two different ports? if there is any other option we can use, plz inform me about it. Thanx in advance. -- Best Regards Rizwan Hi...

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 I...

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)?

...what you suggest, how does > it work? > Well I think it's enough of questions for now. :) We did live video and audio streams of the LAC2005: http://lac.zkm.de/2005/ The wiki is still up: http://footils.org/cms/pydiddy/wiki/LinuxAudioConference2005 I've continued using a similar settup to stream meetings hosted here at RFA. I'm doing another one tomorrow evening. Basically I get digital video from a camcorder via ieee1394 into a laptop. Inside the laptop my toolchain is: dvgrab | ffmpeg2theora | oggfwd It works fairly reliably. I'm looking for other options, though, so...

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

...r user in order for this to work. Since the verion of openssh 3.4 and above came i cant seem to ( so far ) hack this together to get this working. Not sure if the preveldge separation had anything to do with this or not. I have noticed that on Redhat this isnt an issue, displays are able to be settup correctly no matter how many times you su to different userid's the display can be issued back to your desktop. Have you noticed this on AIX ? Do you know of a way to correct this on AIX ? Is there anything more i can do ? I've even got IBM working on this since they now are prov...

...message when Asterisk issue a INVITE without authentification. However, after receiving the 401 Asterisk doesn't send back another INVITE with credentials. Why this happen? When I call from the outside my number at addaline, asterisk answer and send me to my soft phone without any problem. My settup is something like this: sip.conf [proxy.addaline.com] type=friend host=216.87.144.203 username=5555555555 secret=secret context=from-sip [2203] type=friend username=2203 secret=secret host=xx.xx.xx.xx mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 extension.conf: [from-sip] exte...

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.ge...

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...

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

...at bubba3-one:/etc/sssd# host -t SRV _kerberos._udp.earth.local _kerberos._udp.earth.local has SRV record 0 100 88 bubba3-one.earth.local. root at bubba3-one:/etc/sssd# host -t A bubba3-one.earth.local bubba3-one.earth.local has address 192.168.1.1 root at bubba3-one:/etc/sssd# I am now trying to settup sssd using https://wiki.samba.org/index.php/Local_user_management_and_authentication/sssd sssd seems to start fine (no errors in the log and the daemons are running), but getent passwd and getent groups returns nothing. Below is my config: [sssd] services = nss, pam config_file_version = 2 domain...

...so on should be right. The attachment is the output of "pri debug span 1" from Asterisk-CLI, making a call from extension 66 to an Austrian number. - We cannot determine where the leading zero is added to the caller ID. - Is there another way to debug this? Not from asterisk-CLI. Our settup is the following, Asterisk 1.2.19, zaptel 1.2.18 Digium TE120P T1/E1 card -> zaptel.conf: loadzone=nl defaultzone=nl span=1,1,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 -> zapata.conf [trunkgroups] [channels] language=en context=isdn-incoming-e1 switchtype=euroisdn ;internationalp...

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@defaul...

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:1...