similar to: Status of latest round of Allison recordings

Hello - I am going to have Allison record some "intro" messages in my next round of text to her. I'd like to have a consistent voice during the first messages at a company, even if the user presses buttons that take them to a different voice for the alternate language. In other words, I'd like to have Allison say: Thank you for calling Foo, Incorporated. Para

Thanks to a generous donation by VoicePulse (http://www.voicepulse.com/) there are now a wide selection of useful (and not-so-useful) voice prompts available on my website, recorded by Allison Smith (http://www.theivrvoice.com/) There were some apps and dialplan tricks that people were waiting on release because of a lack of voice prompts; hopefully I received and remembered your requests.

I have Meetme2 (as well as the web ui) installed and am having some difficulty with the admin features. I've set up two extensions pointing to the same conference, one with the admin flag (1234|Maps) and another with (1234|Mmps). My issues: If the admin presses the * key, it goes to an endless loop of "enter the conf no. followed by pound key" If the user presses the * key, the

I'm sending in a set of voiceover requests to Allison Smith this afternoon. I haven't kept up with the -users list to know if there is someone keeping track of this stuff any more... We only have a few phrases for her to record, and if anyone has applications which require Allison's voice for the "asterisk-sounds" repository, let me know. I'll be sending this in

I was searching for help on how to handle the errors that are returned from the MeetMe application. for instance. 1) if a user tries to join a conference that is locked, allison says that the conference is locked and then comes back to the dialplan, however it does not continue down the dialplan. I have a meetme command on Priority 8, and the CLI says that it returned non zero (as the wiki

Hello list, I wonder if someone might be able to clear up something for me. I recently set up asterisk and have now managed to get the MeetMe application up and running. When I dial the extension to access the conference/MeetMe application, the only prompt I hear is:"You are currently the only person in this conference." When I use a friend's newly installed asterisk, I hear:

Hi, Allison Smith continues to contribute to the open source asterisk resources and she is launching a new site that will make it even easier to grab sound files. Allison joins us to talk about that and whatever else comes up. She says: "I have been the Voice of Asterisk -- the world's fastest-growing telephony platform -- since its inception (which for me, was marked by an animated

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

So, per the discussion last week and generous donations, we have some new sound files with which to work. The sounds are located in: http://www.loligo.com/asterisk/sounds/ For those of you who just want to download the _new_ sounds, please fetch: http://www.loligo.com/asterisk/sounds/20040117.newsounds.tar All of the sounds in that tarball are also in the main ../sounds/ directory in

Hi, I am translating the voice files of voicemail now. I don't know if the POUND and # are the same key in the telephone's keypad. If they are same, how could we understand the following message: %vm-msginstruct.gsm%To hear the next message press 6, to repeat this message press 5, to hear the previous message press 4, to delete or undelete this message press seven, to quite voicemail

Hi, I know this question may have been asked before (although the archives don't seem to suggest it), but has anyone had any problems with Asterisk accepting a PIN number for a conference room. At this point in time I have established the conference definition in the meetme.conf file as well as specifying the appropriate lines in the extensions.conf file. meetme.conf file: conf =>

Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: "Thank you for calling xxxx". I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the "phpconfig for Asterisk PBX" page, and selected

I am using Festival to synthesize some menu Interaction with a caller and am having a problem. What I am working on is a remote callback where I can remotely call in to an extension, and enter a callback number (or use the CALLERID info) and a second outbound dialing number to connect to. Things work O.K. until I set usecache=yes in festival.conf. After doing this, things run well for

Hello, I've been working on an * dialer application, whereby a requirement is that if no one answers the call, a message must be left on voicemail. I've been using the record(tmp.gsm) function with silence detection enabled to wait for the greeting to finish before speaking. However, on voicemail systems where you can interrupt the greeting with a pound (#) key to access your voicemail

Due to changes at the office, I'm finally getting around to setting up an AA to deal with incoming calls. One of the big changes is that we're dropping the old alphanumeric pager and will just send pages to our phones. I've got the outbound greeting message working in a test context no problem right now, but I'm kind of stuck on how to capture a DTMF sequence from a user and

I'm running Asterisk 1.8 from Debian. I have some analog phones connected via a TDM400P. I'm testing them with these simple extensions: exten => 600,1,Answer() same => n,Festival(This is an echo test) same => n,Festival(Hang up or press pound when you are done) same => n,Echo() same => n,Festival(Good-bye) same => n,Hangup() exten

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *

Hello, So far everything that I'm trying with asterisk is working except for this weird thing. When I try to call voicemail and it asks me for the password I enter it in but from the debug message I can see that it thinks I didn't enter anything in. Also when I'm leaving a message it sais press pound to end, but even if I press it 10 times it keeps on recording until I hang up. It

Hi, Did someone experience that Asterisk OS 1.2.5 voicemail issues? Problem description: Some one call to the extensions 200, After 10 sec ring then go to voicemail 200@default. Announcement "Please leave me a messages.....blar blar.." When I completed to leave a message... IF : I press the pound "#"key ... Then it says "Transfer" IF : I Press the zero

Hi, i want to to this thing with php AGI: #!/usr/local/bin/php -q <?php set_time_limit(30); require('phpagi.php'); error_reporting(E_ALL); $agi = new AGI(); $agi->answer(); $cid = $agi->parse_callerid(); $agi->text2wav("Hello, {$cid['name']}."); $agi->text2wav('Enter some numbers and then press the pound key. Press 1 1 1 followed by the pound key