similar to: Re: [Asterisk-Dev] important feature missing?!

Hi, Asterisk has a lot of customizable voice prompt in /var/lib/asterisk/sound but i want to change a very well known voice message which occurs when we try to dail a number against dial plan "beep beep beep The person you are calling is unavaiable, please try again." I thought it would be availabe in the sound directory of asterisk but it is not there. When i dial such wrong number no

I have a voicemail application that users can listen to messages and leave messages. I am looking for a way to play a beep tone to a user when a new message is received when they are on the phone. Here is what I have come up with: in extensions.conf: [beepvoicemail] exten => 1000,1,answer() exten => 1000,2,NoCDR() exten => 1000,3,wait(2) exten => 1000,4,Set(TIMEOUT(absolute)=5)

Hi, has the following been done before respectively is it possible with Asterisk? I searched the archives but couldn't locate anything. 1. Call to 5555 comes in via SIP. 2. Call is not answered yet but progress continues. 3. At the moment the call comes in something like this gets spawned in the background: Dial(SIP/123456 at provider,,D(ww${EXTEN}) which should translate to:

I spoke the other day about my preliminary tests with office-wide paging with Cisco phones using the new SIP 6.1 image which supports auto-answer. I've got a small and crude recipe for those of you who want to experiment and hopefully create some better and more complete examples than the one I've thrown together below. Create a new line on each of the Cisco phones, and put the

AGI's are resulting in unusual behaviors. Can someone please tell me if this is my inappropriate use of AGI's, inappropriate use of Time::HiRes, or a bug with *: I call this script twice: #!/usr/bin/perl use Time::HiRes qw( gettimeofday ); ($seconds, $microseconds) = gettimeofday; $hirestime = sprintf("%s","$seconds$microseconds"); print "SET VARIABLE

Hello There, I would like to play a busy tone (ie BEEP BEEP BEEP) when the underline carrier sends back 486 Busy Here. Looking at Dial parameters ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial), it mentioned something about the r parameter as not being very professional or something like that... Then there was: U(x): Executes, via gosub, routine x on the called channel. This is similar

The soundfiles I submitted earlier today have been cleaned up, and added to the Digium CVS server in a more formal manner. Also, some of the really bad formatting in my .txt description file has been rectified. All of the sounds on my website are now on the Digium site, and I will be submitting future changes via patches to Digium for additional sounds. Ideas welcome for more text; I may

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients(using linphonec). In a proper context, I have mentioned extensions 107 as simputer@X.X.X.X (x.x.x.x=asterisk server ip) Asterisk Sever-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients. In a proper context, I have mentioned extensions 107 as simputer@bogus.com Asterisk Server-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on asterisk terminal ---------------------

There are no cordless devices in this situation. :) Jerry Geis wrote: >/ I have a normal setup of calls coming in on analog lines (4 of them) />/ coming into />/ an old KX1232 pbx. I have those lines forwarded to a T1 card in the />/ KX1232 />/ going over to my T1 card, it then uses IAX going over to my REAL />/ asterisk pbx. />/ />/ (these steps are there for testing

Hi, [I'm Using redhat linux 7.1 with R-1.4.1 rpms.] Last summer, Henrik Bengtsson wrote to R-help: > Is there a way to turn of the (annoying) beep that occurs when one calls the > locator() command and clicks the mouse. Try > > plot(0) > locator(n=10) > > I am running Rterm on WindowsMe/Cygwin bash and I don't know if this is the > case on other platforms.

The freenum.org project wants to use your trunks! The freenum.org project is an ENUM parallel tree, which has as an eventual goal the distribution of ENUM numbering in nations or areas which due to political or other issues are not able to get secure, inexpensive, or functional ENUM capabilities. As a preliminary round, we're putting toll-free gateways into the system for various nations.

Hello all, I've played with background and play sounds apps and googled around and asked the list before to no avail. Does anyone know of a way to have tones played during the call progress stage of the call? We (especially on some international circuits) get up to 5 seconds of silence before the phone starts ringing or is busy. I don't want to force "R" on the Dial app as

So, just today I noticed a problem with kernel 2.6.18-194 (CentOS 5.5) on several Intel DP965LT systems; the system beeps (such as terminal beeps) are no longer passed through to the external speakers. This is a problem because in our situation the boxes are distant from their monitor/keyboard, the system speaker on this motherboard is extremely weak, and there are no system speaker header pins

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re:

Asterisk 1.2.4 FreePBX 2.0.1 I am running a TDM400 card with 3 FXO and 1 FXS cards. During most all calls, there are random beeps in the background. The other party cannot hear this (I believe since they haven't said anything). This will happen when people leave voicemail, and also on most or all calls. I am running Polycom IP 501 phones. The beep/tones are not DTMF. It does not happen

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what

Hello, I would like to have "whisper" channel spy (not private whisper) in Asterisk 1.2. I see here: http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html That is only available for Asterisk 1.4. I wonder if there is any way to emulate it in Asterisk 1.2. For example, to "join" two calls: one to use a private whisper and other one to use a normal chanspy.

Any ideas on how to disable the audible incoming call beeps with *. We have a dial pool and if someone calls into the pilot number and we're talking to them, if another call comes in, both us and the far end users hear the incoming call beep. -------------- next part -------------- An HTML attachment was scrubbed... URL: