similar to: Voice prompts, do we have to use GSM?

Hi We are trying to get better sound quality out of the prompts on our Asterisk system. We had some new ones made by thevoice.digium.com and they are in WAV format instead of the default GSM format on the Asterisk server. The problem is, when you dial in to the server using the FXO card (X100P) you really can't tell the difference between the WAV prompt and the GSM prompt, however, if you

Repost, first one never made it to the list... -----Original Message----- From: Gregory Wiktor - ADCom Corp. Sent: Thursday, September 29, 2005 3:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Voice Prompts, what do you think? Good voice. Should we record a new prompt-set? Hello all, I have someone working for me who has a nice phone voice. I looked at

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've spent the past week experimenting with Asterisk@Home 2.6, and then Asterisk 1.2.6 individually, on VMWare Workstation 5.5. I have an entirely IP (hard & soft)phone setup (IAX and SIP) so I have no requirements to support any Digium PCI cards, etc. All in Asterisk works extremely well except for one thing: Playback of sounds (GSM format) such as an ivr greetings, sound terrible.

Hello, after upgrading my asterisk from cvs few days ago I discovered that all my anddefault asterisk voice messages/prompts was played wrong - they was played so slowly ... and the IVR system wasn't usable at all. Today I upgraded everything: a fresh cvs copy, made make install and make samples - but after that the problem with IVR was the same: ALL default asterisk sounds in gsm format

Hello all, I have someone working for me who has a nice phone voice. I looked at some available prompts for asterisk, and found both the free and commercial ones to be pretty horrible. The asterisk ones are good, but I wish I had more to choose from sometimes. My question is, what do you think, should I bother having her record a full spectrum of prompts for asterisk? If you want to hear her

Hi list, I just downloaded the ulaw prompts from this website: http://www.gmvoices.com/solutions/detail/asterisk/ as had a few complaints about the levels and accent of the existing prompts. The downloaded prompts sound great when listened to on my desktop machine, however when I use them on our CentOs 5.2 / asterisk 1.4.21.2 server there is a fairly loud popping noise in between each sound and

I've looked all around, and I can't find an answer to this. I apologize if this has been discussed already or is buried somewhere in voip-info.org. I have an asterisk setup on linux 2.6.11.11 kernel, a revision E/F TDM400P, and Polycom IP501 phones. As soon as I load the zaptel module into the kernel, the voice prompts and voicemail system ceases to work. The asterisk logs

Lets say I have an * at my business, with 7960 SIP phones. All the sip phones are registered using their extension number (like 305), but I would also like to put my SIP URI on my business card and in a name format, not an extension number (like lee.goodman), so that the SIP URI would read lee.goodman@asterisk.company.com. How would I set this up in extensions.conf? I got

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've have installed a new Asterisk 1.4.15 system after having previously used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though the newer one is actually a slower processor. On the new system, playback of gsm files is noticeably poorer (voice quality is flakely) on any connected phone (sip or isdn, internal or external).

Before you get too far.... The internet line jacks dont allow outbound calling. they cannot be used as trunk lines to the PSTN. the outbound code has not been written yet. I had to go buy FXO card from digium (that works much better than the Linejack) to get outbound calling to work Dave >>> fplandae@hotmail.com 4/11/2003 5:35:24 PM >>> Hi, I am a newbie. I have been

When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing to be transcoded on the fly and it's not getting transcoded fast enough. I did a file convert to create gsm versions (currently they are referenced in my dial plan with no extension Seem to have same problem. How do I determine which file

I am, as this question will make very obvious, not exactly a guru when it comes to audio under Linux. I'd like to record some prompts using a nice microphone and my Ensoniq sound card, and then convert them to gsm/wav for use with asterisk. What Linux command would I use to a) do the recordings, and then b) convert them to the appropriate formats? I know there are

I am using Audacity to record some voice prompts. The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. I know that some of this comes with the territory, but I wonder if there is anyone out there who does this routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting

Hello- What is the .gsm format? Ie: what's the encoding method and sample rate please? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott@evtmedia.com URL: www.evtmedia.com

Hello: I would like to produce some GSM Prompt audio files for a Telephone Directory Project-- and have hired a freelance audio engineer to record, and edit the actual files-- However the GSM files he gives me to upload into asterisk DO NOT work when played back throgh "Stream File" or "Get Data" in my agi. It seems that there may be more than one GSM file type (with

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

Hi, When listening to the time and date a voicemail message was received, you can hear french sentences like : "message re?u ? vingt-et-un heure" (message received at twenty and one hour) It should be (in whatever french flavour): "message re?u ? vingt-et-une heure" In short, instead of: -- <SIP/7533-094fdbb8> Playing 'digits/at.gsm' (language 'fr')

Last week I was able to do some debugging of the problem I'm having with IAX2/GSM, residential-grade broadband, and VOIP. To summarize, I am having a great learning experience with * and Zap cards, SIP and IAX2. I hit a wall though, when I registered with iaxtel and tried doing VOIP. I spend the better part of a workday with the jitterbuffer and all sorts of settings and finally started to

Hello: Thanks for the input so far. Heres the issue-- This is a production environment-- where many people "touch" the files. ie-- The audio engineer is a freelancer who wants to master the files at the highest quality TO HIS EAR and experience-- He knows NADA, Not a thing about SOX-- but is a ProTools GURU. The SOX resampled files work on our asterisk box-- but I gotta put someone