similar to: issue with the detection of the call status after sending it using Orginate (DAHDI/1/...., app, ...

hi evrey one, i'm in some kind interesting in developping some asterisk programme like doing a small programme including some of these services that do a telephone operator. but abviously i need to know about programming in asterisk in thos to files i think :) (extensions.conf and in sip.conf files) so i'm asking if someone can give me a puch,i will be very glad thanks in advance

Hi, I started playing around with a function for using StatTransfer (version 10) for importing data. This started as a simple task but it's not working and so now I'm very frustrated. I'm using R version 2.13 on Windows 7. The function, called fn.importData, is: function(file = NULL, type = NULL){ ## ## create statTransfer command file -

Hi every body; I want to intall some softwars working with my Asterisk server and I get these erreurs : * error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. /usr/sbin/grub-probe: error: cannot seek `/dev/sda'. dpkg: error processing grub-pc (--configure): subprocess installed post-installation script returned error exit status 1 *

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

Hi there, I have an issue , I want to make a video call to a streaming source using Asterisk . Someone can help me in this issue please? Thanks in advance. -- *?l?ve Ing?nieur INE3 ? l'Institut National des Postes et T?l?communications * *INPT - Rabat - Maroc* *Responsable de la cellule Asterisk au **Club Electronique et Systemes Embarqu?s de l'INPT* *Membre du projet "

hi everyone, actually i want to release an IVR system using PHPAGI API , in this IVR i want to get value from the user. I already used get_data defined in phpagi but they are not able to get the value given by the user and store it in a php variable. i tested this : $result = $agi->get_data('beep', 3000, 20); $keys = $result['result']; but every time i found in $keys variable

I have a list of txt files that I want to convert into .rdata R data object. filenames 1. "./file1.txt" 2. "./file2.txt" 3. "./file3.txt" 4. "./file4.txt" 5. "./file5.txt" 6. "./file6.txt" 7. "./file7.txt" 8. "./file8.txt" 9. "./file9.txt" 10. "./file10.txt" I saved these files as for ( i in

I am trying to install the binary version of xen 3.1. When I boot, after some console output, I get the following: :: Loading Initramfs scsi_mod: no version for "struct_module" found: kernel tainted. SCSI subsystem intialized :: Running Hook [udev] :: Loading udev...input: AT Translated Set 2 keyboard as /class/inpt/input0 done. :: Running Hook [keymap] :: Loading keymap...Cannot find

Greetings list, Have there been any further developments recently regarding presenting the original caller's caller ID to SIP devices after an attended transfer? I've googled around on the topic, but most of the threads I've found (some from this very list) are all dated back in mid-2006 and I wondered if there have been developments on the issue. To recap, the desired behaviour

--- Matt Zimmerman <mdz@debian.org> wrote: > retitle 119396 flac: Decoded raw data does not match original raw > data > thanks > > On Tue, Nov 13, 2001 at 02:38:32AM +0000, Adam M. Costello wrote: > > > Matt Zimmerman <mdz@debian.org> wrote: > > > > > Based on a few quick tests, this kind of problem seems to occur > with > > > any

I have an urgent need to install the original Icecast (1.3.12). I have a piece of broadcasting software that can only receive a stream from the "old" version of Icecast. I've tried building it under Debian 9, but failed. Can anyone provide any assistance? I'd prefer to stick with Debian, but I'm willing to use a different distro if necessary. I've worked in IT for

retitle 119396 flac: Decoded raw data does not match original raw data thanks On Tue, Nov 13, 2001 at 02:38:32AM +0000, Adam M. Costello wrote: > Matt Zimmerman <mdz@debian.org> wrote: > > > Based on a few quick tests, this kind of problem seems to occur with > > any file smaller than 4k (at least when using 44100Hz/16bps/2chan). > > I don't think the problem

retitle 119396 flac: problems encoding very small files thanks flac-dev: I received the following bug report from a Debian user. Please preserve Cc: 119396-forwarded@bugs.debian.org when replying. On Mon, Nov 12, 2001 at 05:44:09PM -0800, Adam M. Costello wrote: > Package: flac > Version: 1.0-3 > Severity: grave > > The file "test" is a 1000-byte raw audio file (44100

Hi Bilal, if you're looking for Asterisk, CRM and even Google Contacts integration, do check out Aptus FonB (www.aptus.com). I believe that's the exact solution you're looking for. Br, Maax __________________________________________________________________ Hi; I used vicidial for call center and I would like to try elastix. Can someone advise about the advantages? Does Elastix

Hello SUN gurus I do not know if this is supported, I have a created a zpool consisting of the SAN resources and created a zfs file system. Using third part software I have taken snapshots of all luns in the zfs pool. My question is in a recovery situation is there a way for me to mount the snapshots and import the pool while the original is still active. Right now all I am able to do is export

Hi All, I'm trying to use the orginate cmd. I have it working if originate is from a user e.g. SIP/7777 originate SIP/7777 extension 987654321 at outbound-route What i'd like to be able to is instead of a local extensions i would call an outside number then connect it another outside number. e.g. originate SIP/85431210 at outbound-route extension 987654321 at outboudn-route is this

Hello all, Here is the requirement from my side to use Asterisk.NET API to generate an automated call (outgoing) from asterisk and then link to one of the extensions which plays a sound file for the callee. For this i have worked out in the follwing way 1)modified manager.conf to facilitate this API to talk to asterisk 2)used the command Originate to call a Registered user under

Hi, I have only one ethernet port in a remote server. (eth0) I have a public address with x.x.x.164 netmask 255.255.255.240 gw x.x.x.161 and want to use in my guest OS the next available ip address (x.x.x.165 netmask 255.255.255.240 gw x.x.x.161) Is this posible with brctl to achieve this? I did a file called ifcfg-xenbr0 with: DEVICE=xenbr0 TYPE=Bridge BOOTPROTO=dhcp ONBOOT=yes then

Hi, I have only one ethernet port in a remote server. (eth0) I have a public address with x.x.x.164 netmask 255.255.255.240 gw x.x.x.161 and want to use in my guest OS the next available ip address (x.x.x.165 netmask 255.255.255.240 gw x.x.x.161) Is this posible with brctl to achieve this? I did a file called ifcfg-xenbr0 with: DEVICE=xenbr0 TYPE=Bridge BOOTPROTO=dhcp ONBOOT=yes then

I've got a piece of CPE equipment that has an FXS port that I have tied to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a standard telephone connected to it, I get a dialtone. If I dial a digit, and send a hookflash, the device will provide a dialtone back for the next available channel on the device. I'm trying to recreate this same behavior with Asterisk,