similar to: Playback uses channel's language, background doesn't

HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has

Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound

Hi, There seems to be a problem with setGroupGeneric() from package 'methods'. The symptoms are somewhat erratic, in the sense that small changes may lead to any of the following behaviours. 1.Package works without problems on Windows and installs on Linux but gives error when loaded with library(). 2.Package installs on both systems and gives error when loaded on any of them. 3.

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure

I have a dialplan that looks like this: [dorecord] exten => _*99XX,1,Verbose(2,Doing custom record) exten => _*99XX,n,Answer() exten => _*99XX,n,Verbose(2,Doing custom record - before wait) exten => _*99XX,n,Wait(0.5) exten => _*99XX,n,Verbose(2,Doing custom record - before record) exten => _*99XX,n,Record(/tmp/prompt${EXTEN:3}.gsm) exten => _*99XX,n,Verbose(2,Doing custom

I am hoping someone else has seen this and can offer a solution or at least a direction to investigate. I am running 11.23. Most of my clients are fine but one has a strange behaviour. He has a Grandstream HT701 like most of my clients who use an ATA. He can make call and they are crystal clear. However, when he tries to use phone menus ("dial 234 for John Doe" for example) it

My rails log get chock a block with things like <<<<<<<<<<<< ActionController::RoutingError (No route matches "/admin/Y- ivrrecording.php" with {:method=>:get}): ... ActionController::RoutingError (No route matches "/manager/html" with {:method=>:get}): .. ActionController::RoutingError (No route matches "/webdav/test" with

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi, I finally got Asterisk installed using the German installation CD from http://www.asterisk.de.ms. I got two SIP phones working (SIPPS) asterisk*CLI> sip show inuse Username incoming Limit outgoing Limit 5678 0 N/A 0 N/A 1234 0 N/A 0 N/A

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Hi R community, I have been trying to figure out why R is reversing the order of rows after I run data.matrix() Here is my data: df<-structure(list(itmID = c(1L, 2L, 1L, 2L, 1L, 2L), variable = structure(c(1L, 1L, 2L, 2L, 3L, 3L), .Label = c("3", "2", "1"), class = "factor"), value = c(0.7, 0.52, -1.16, -0.91, 0, 0)), .Names = c("itmID",

What ? ====== littler - Provides hash-bang (#!) capability for R (www.r-project.org) Why ? ===== GNU R, a language and environment for statistical computing and graphics, provides a wonderful system for 'programming with data' as well as interactive exploratory analysis, often involving graphs. Sometimes, however, simple scripts are desired. While GNU R can be used

Hi folks, after upgrading a Centos container (LXC) from 7.4 to 7.5 I got this: # systemctl status Failed to get D-Bus connection: Connection refused # busctl Failed to connect to bus: Connection refused # ps -ef | grep db[u]s dbus 55 1 0 Jul02 ? 00:00:00 /usr/bin/dbus-daemon --system --address=systemd: --nofork --nopidfile --systemd-activation # lsof -p 55 COMMAND PID

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Hello all.... I need help and I'm just about at my wits end. First let me state that I can not give logfiles in email since the Samba servers are on the secure side of my world and contain sensitive information. If by my stating my problem, I am hoping that someone will point me in certain directions that I have not tried. We have two samba servers on our site. Both are running Samba 3.0.0.

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,