similar to: saydigits in another "language"

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

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

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 all -- I'm having awesome fun with Asterisk & voicepulse connect together. So cool. I'm trying to have the caller id read back to me. Do I need to do something to have this sent across in the sip.conf? Or is there something I need to do somewhere to enable the reading of this data? Thank you! Matt Here is my extensions.conf exten => _XX.,1,Answer() exten

I'm Japanese. Sorry,English is not so understood,Please let me question by items. In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of Authenticate application's 'j' option. exten => 123,1,Answer() exten => 123,2,Authenticate(789,j) exten => 123,3,Playback(pin-number-accepted) exten => 123,4,SayDigits(111) exten => 123,103,SayDigits(999) In this

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 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

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

After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root at localhost asterisk-11.1.2]# asterisk -vvvvvvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components

Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file:

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:

Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g but every time I try to play something I got: -- Executing [301 at from-test:1] Dial("SIP/300-00000045", "SIP/301,60,rjtTg") in new stack -- Called SIP/301 -- SIP/301-00000046 is ringing

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

I have put together this call forwarding and dnd config: I'm sure it can be dome with macro's but I couldn't figure that out... anyone care to input. 74 Turns DND on my phone will not ring, drops caller to voicemail... 73 Turns DND off 72+ext forward your extension to another extension and voicemail is left at the forwarded extension. 71 turns off call forwarding. ; dnd Could

Guys Im having a few issues with Languages. Ive setup the english language is it came from default: /var/log/asterisk/sounds /var/log/asterisk/sounds/phonetic /var/log/asterisk/sounds/digit /var/log/asterisk/sounds/letters and then Spanish as /var/log/asterisk/sounds/sp /var/log/asterisk/sounds/sp/phonetic /var/log/asterisk/sounds/sp/digit /var/log/asterisk/sounds/sp/letters Then one of

I need to dial an extension that tells me what extension I'm dialing from. I'm running a bunch of analog phones off a channel bank to * over a T1. I have the following in extensions.conf. exten => 98,1,SayDigits(${EXTEN}) This says the digits the caller enters on the keypad, not the extension they are calling from. Thanks Guys!!!!!!!! Paul Paul Mahler pmahler@signate.com

I get the following message when I dial 74#. Does anyone have any ideas on what might be going on? If I don't require numbers to be terminated with # everything works as expected, but you have to wait for the digit timeout, of course. MESSEGE: DEBUG[1150520624]: File pbx.c, Line 1683 (ast_pbx_run): Oooh, got something to jump out with ('#')! -- Invalid extension '#' in

Here is the current dial plan section: [custom-michael] exten => _900,1,Playback(custom/extn-xfer) exten => _900,2,SayDigits(${EXTEN}) exten => _900,3,MixMonitor........... exten => _900,4,Dial(SIP/${EXTEN}|${DEFRT}) exten => _900,5,Playback(custom/extn-xfer2) exten => _900,6,Goto(custom-michael,901,4) exten => _901,1,Playback(custom/extn-xfer) exten =>

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.