similar to: SayDigits

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 =>

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

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

Hello I need to make site views changeable I found how to change layouts but it is not enough, becouse templates can use different ways to display and organize content. I found one way to do this, but i need to make subfolders in my ''view/layouts'', ''view/controller_name'' both and to create additional subfolder somewhere in ''public'' to put

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've got this on the backup DC root at bdc:~# wbinfo --sid-to-gid S-1-5-21-1166961617-3197558402-3341820450-516 3000000 while root at bdc:~# ldbedit -H /usr/local/samba/private/idmap.ldb objectsid=S-1-5-21-1166961617-3197558402-3341820450-516 shows correct xid 3000019 and on the primary DC I've got itk at dc:/$ wbinfo --sid-to-gid S-1-5-21-1166961617-3197558402-3341820450-516 3000019

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

Hi, But in fact, it really affects the voice quality. One of my tester says, "Is your mouth far way from the mic?" Could you explain why we should cut 200hz below? >The notch filter is specifically designed to cut below 200 Hz when >working in narrowband. In wideband, the cutoff is more around 50 Hz. The >reason is that in narrowband operation (irrespective of the

Hello I have one question about cluster and vmware. I have about 10 computers they are all old once from 1g and 256 ram. And now im thinking of putting them all toghter in one cluster. This cluster should be an high performing so all computers share they performance. Now on this cluster i whould like to install one vmware server and make that vmware server get the power from all these

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

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 am developing an ACD front end using Asterisk 1.2.14. I heard that AgentCallBackLogin will be deprecated in future version of *. Is this true? If it is, how can I use AddQueueMember to replace AgentCallBackLogin? I mean to login an agent in multiple queues at once. I have multiple queues and a lot of agents defined in queues.conf and agents.conf. Each agent may login more than one queue. It

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

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

I've been looking around for an example of a method of reading back a caller ID value, but I haven't found anything that doesn't use Festival. I'd rather not resort to the Mr. Roboto voice if I can avoid it. Playback of the numbers one at a time is perfectly fine, so I'd like to use the default female Asterisk voice (the sound files are in place on my server). Does anyone have

Greetings *`s, I am having what appears to be a small problem, but the frustration is erally getting to me, what am I doing wrong here ? I used AMP to set up a custom menu, so if caller presses 1 it goes to ext200, if caller presses 2 it goes to ext201 etc etc... Now I have created a third option that when the caller presses 3 it must play a sound and hang up. No rocket science yet. When

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.