similar to: Interrupt Asterisk's SayDigits()

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 have a user who wants to receive an ANI spitback in DTMF. Right now, the "SayDigits(${CALLERIDNUM})" command works fine with voice. But I'd like to end up doing both. Something along the lines of: exten => 34,1,Answer exten => 34,2,Wait(1) exten => 34,3,Playback(vm-extension) exten => 34,4,SayDigits(${CALLERIDNUM}) exten => 34,5,Wait(2) exten =>

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

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

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

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

On Fri, 2007-07-20 at 02:08 -0400, BSumrall wrote: > My dial plan of issues?.. > exten => s,1,Answer(60) > exten => s,2,Background(otherwise-press) > exten => s,1,Playback(digits/1) > exten => s,2,Goto(default,s,1) > exten => s,1,Playback(digits/2) > exten => s,2,Goto(default,s,1) I'm not sure why you have three different sets of priorities one and two

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

Hello, I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk 1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine with SIP channels without Festival. I have written following context in extension.conf: [connect-to-me] exten => s,1,Answer Exten => s,n,SayDigits(?1?) exten => s,n,Festival(hello john) exten => s,n,Hangup I use call files to

Markus Weiler <markus_weiler at mailworks.org> schrieb: Hi > from voipinfo... > > If an Asterisk command specifies a sound file in a*subdirectory*, > Asterisk looks in that subdirectory for the language subdirectory. For > example, theSayDigits > <http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigits>command may > play the sound file

Hello, Suppose a dialplan such as: exten => 6004,1,Answer exten => 6004,n,Wait(1) exten => 6004,n,SayDigits(1) exten => 6004,n,Festival(This is a test of Festival) exten => 6004,n,Hangup When watching in the CLI, I see this: == Using SIP RTP CoS mark 5 -- Executing [6004 at internal:1] Answer("SIP/505-00000004", "") in new stack -- Executing [6004 at

Our setup: We have a DS3 from Global Crossing terminating into a Adtran MX2800 M13 mux. From there, groups of 4 T1's run into T410P digium cards to 7 individual servers. Each trunk is configured as ISDN PRI, B8ZS/ESF, D-channel being chan 96 with B-channels of 1-95 (we're using NFAS). The D channel is up and there are no alarms. We see the connection on the console from the

Hi all, I'm unable to put app_festival to work. I successfully patched, installed and tested festival (interactive logon and telnet to server port) which seems to work without problems. But when I test it in asterisk I got the following trace in console: -- Executing Answer("SIP/bsenicar-850b", "") in new stack -- Executing

Hi all, I just installed Cepstral and app_swift version 1.4.2 on my Asterisk 1.4.22.1 box. It seems to work great with one exception. If I play a test message with instructions to collect a maximum of 5 digits, it collects those 5 digits correctly if the user waits for the message to complete before entering them. But if the user barges in with digits before the message completes, the detected

So, We want to be able to put a fax machine on the line port of the X100P in our asterisk server. We however also want to use this card for 911 calling. We need some sort of mechanisim to "disable" the line out port on the x100p by software to "interrupt a call" on the line. Anyone done anything like this?

On 1/11/07, Mike D'Ambrogia <miked@jamagination.com> wrote: > > Ralph > > Kind of new to asterisk, and really new to AGI but it looks like you were > trying to have the AGI script tell asterisk to read and lay the results into > my_var and then regain control in the AGI script, is that correct? > > If so I don't think that will work since the dialplan

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK

Hi, I'm playing with the agent/queue system. Everything work well with v1.0.3. but I want the 'Action: Agents' in the manager API that is only on the CVS version. So i switched to, but now the Queue/Agent system barely work. (my agent don't get the call) Where I can get a 'stable' CVS version? Or maybe, how I can solve my Queue/Problem? here is the detail: 1. I can