similar to: SayDigits -- ToneDigits??

Listonians, Anyone get * to work together with Vonage? Thanks, Jerry

Dear R gurus, I regularly come across a situation where I would like to apply a function to a subset of data in a dataframe, but I have not found an R function to facilitate exactly what I need. More specifically, I'd like my function to have a context of where the data it's analyzing came from. Here is an example: ### BEGIN ### func<-function(x){ m<-median(x$x) if(m > 2 &

Hi. I need help with my brand new Voicetronix OpenLine4 board that I installed into Asterisk. After building the Linux device driver and inserting the module, I modified the /usr/src/asterisk/channels/chan_vpb.c file to uncomment the US settings and comment out the Austrailian ones. I made the appropriate entries for routing in vpb.conf and extensions.conf.... All appears to be well, except

One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp

Is anyone successfully using the AGI script calleridnamelookup.agi (or anything similar) ? I get both name and number caller ID from my POTS line, but I'd save money if I had them deliver ANI only. I've downloaded and installed the AGI script calleridnamelookup.agi, but I always get -- Executing AGI("SIP/9525485560-5359", "calleridnamelookup.agi") in new stack

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can

Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring

Hi there, Since last 2 days iconnect's incoming is not working. Is it the same with everybody? For the past 5 months I've been using this service perfectly in two boxes and suddenly it stopped functioning. I'm able to call out, the version is 0.9.1. Any help is appreciated Thanks, Raj --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB

I've been trying to play the default music on hold file, but no luck yet. here is my configuration: extensions.conf [incoming] exten => s,1,Dial,Zap/2|10 exten => s,2,Voicemail,u34 exten => s,102,Voicemail,b34 exten => 34,1,SetMusicOnHold,default Musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random =>

Has anyone done a modification where you can Interrupt Asterisk's SayDigits(). This will be helpful in order to be able to interrupt an announce and dial digits without waiting to hear all the announcements. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081009/06185059/attachment.htm

Hi, I am unable to get any music or sounds played with the MusicOnHold or SayDigits commands. I do get sound from the Playback and Background commands. I have gone through the process of installing mpg123 and putting the link in usr/bin (and usr/local/bin). For the MusicOnHold command I can see the call come into * and the command get executed I just get no sound on the phone. The * console

I want to rerecord the "1" "2" "3" ... "0" sounds, but not overwrite the defaults. So, I've recorded them into a custom directory /var/lib/asterisk/sounds/custom I was hoping to be able to do the following: exten => foo,1,Set(CHANNEL(language)=custom) exten => foo,2,SayDigits(1234567890) however, I get no errors, but still get the default

Hello Is there a way to slow down or speed up the speed at which SayDigits rattles off a series of digits? Reagards

I am trying to modify a fairly complex digital receptionist dialplan that has a number of included contexts. Right now the system is not announcing the extension that the caller attempted to dial, so callers get confused when they think they dialed a valid extension but asterisk didn't pick everything up. I would like to have the system announce the entension that they attempted to dial in

Any chance of say digits being extended to recognise "*" & "# " ?? Heck these are digits on a normal keypad :-) Gary .

On Sat, Jan 7, 2017 at 2:30 PM, Peter Moody <mindrot at hda3.com> wrote: > so I spent a bit of time looking at this and it seems like the only > way to go, at least if I want to keep it in ssh_connect_direct(), is > to use pthreads. further, it seems like getting that accepted is > something of a long shot: Sorry, pthreads is a non-starter. I would have thought that using

Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

With this, "make syntax-check" runs a few more tests, all passing. >From febff9d2a35c4f40abbaf8943146476bdeac671e Mon Sep 17 00:00:00 2001 From: Jim Meyering <meyering at redhat.com> Date: Tue, 4 Aug 2009 13:49:19 +0200 Subject: [PATCH 1/8] build: remove more files added by ./autogen.sh * po/LINGUAS: Remove file. * po/Makefile.in.in: Likewise. * po/Makevars: Likewise. *

Searching through the archives, I couldn't find an answer for this... I have two asterisk systems, (system A and system B), and would like to use a single voicemail system. Phones on system B are SIP phones, registered at system B. Can the message-waiting indicator be activated on a SIP phone registered to system B, if the voicemail resides on system A? If so, how? Thanks, folks.

Hello! I'm hoping that Gmail won't HTML format this mail so that I'll get flamed :) Anyway, my question relates to ssh_config. The problem I find is that the Host pattern is only applied to the argument given on the command line, as outlined in the man page: "The host is the hostname argument given on the command line (i.e. the name is not converted to a canonicalized host name