similar to: Changing Voice from Male to Female

Displaying 20 results from an estimated 3000 matches similar to: "Changing Voice from Male to Female"

2007 Apr 26
1
Asterisk Voice sound level
Hi, Is there a possibility to control sound levels (higher / lower) in Asterisk (so the codecs). Somebody asked me to evaluate that but I didn`t found any documentation about. I have the opinion that for these (audio) things the end user client is the only part where I can tune around. Problem is for example a (Austria) ISDN --> Asterisk --> SIP / IP ---> (Romania) Asterisk
2006 Dec 20
13
Need quality toll free 800 number over IAX?
Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061220/4919f3cb/attachment.htm
2006 Jun 08
2
Plotting female and male signs
Dear R-users, Just like other users (as seen from previous posts on the list), I would like to use female and male signs in plots. I found B. Ripley's post about using Unicode characters. However, it doesn't works for me. > text(locator(1),"\u2640") produces the following error: Error: invalid \uxxxx sequence But I can specify other Unicode characters as long I
2005 Oct 22
4
Male and female symbols?
Does anyone have an idea of how one might plot male and female symbols on a graph using R? Thanks! .................................................................. George W. Gilchrist Email #1: gwgilc at wm.edu Department of Biology, Box 8795 Email #2: kitesci at cox.net College of William & Mary Phone: (757) 221-7751 Williamsburg, VA
2011 Feb 05
1
Any voice changer applications for Asterisk?
Hello, Are there any other other voice changer applications to Asterisk other than the one from Lobstertech? (http://lobstertech.com/voice_changer.html) Specifically interested in open-source but can have a look at economical commercial alternatives as well. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:
2013 Jul 01
1
Male and female signs as subscript in plot
Hello, I'd like to add labels to my plot that include a male or female symbol as subscript. I'm working in Windows Vista and R 3.0.0. I am able to add the male symbol to the plot as regular text (NOT as subscript), e.g. with: mtext("Male\u2642") This displays the word "Male" followed by the male symbol on the plot. But "\u2642" does not work when I try to
2006 Dec 07
7
Running Asterisk on a Home rotuer
Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm
2002 Oct 31
7
Symbols for male/female
Dear all, I would like to use the biological symbols for male and female as plotting symbols in a scatterplot (ideally filled and non-filled). R does not seem to have these symbols using pch= in plot() nor are they implemented via expression() or at least I did not find them. I found that the symbols are e.g. available in the wasysym and the marvosym package in LaTeX. I have coded two very rough
2002 Oct 31
7
Symbols for male/female
Dear all, I would like to use the biological symbols for male and female as plotting symbols in a scatterplot (ideally filled and non-filled). R does not seem to have these symbols using pch= in plot() nor are they implemented via expression() or at least I did not find them. I found that the symbols are e.g. available in the wasysym and the marvosym package in LaTeX. I have coded two very rough
2006 Dec 27
3
Polycom 601 Contacts List
Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in
2005 May 11
12
Snom 360
I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.
2006 Nov 12
3
Determine if Call is from a cellular phone
Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid -------------- next part
2008 Jun 01
5
New faxing protocol. Good/Bad ?
Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and
2007 Jan 09
2
Attatching VM via email for more than one user
Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in user1@domain.com;user2@domain.com. When the call goes to VM I see in the CLI: uniqueid => 17 customer_id => 0 context => techmast mailbox => 14 password => 1234 fullname => Sales and Service email => user1@domain.com email =>
2007 Sep 18
3
Interesting Conference Request - Can this be done ?
Hi List, I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). Any ideas ? Thanks. Dovid -------------- next part
2007 Dec 11
3
Any phone capable of displaying real time queue statistics?
Are there any phones whose display can show queue statistics, ie: calls waiting, etc, on the phone itself without too much trouble with Asterisk? Especially while the phone is in use (on a call)?
2007 Aug 19
3
Change Packetization Time
Does anyone know if it is possible to change the packetization time in Asterisk ? I was told by a client of mine that adjusting this with using G729 can greatly lower the amount of bandwidth used. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070819/b0cc470f/attachment.htm
2007 Apr 26
1
asterisk slows down when unplugging internet cable with VoIP lines
Hi, I have an Asterisk 1.2.9.1 on a Debian Sarge distro connected to a VoIP provider via internet. I noticed Asterisk gets slow and behaves in strange manner if I unplug my internet cable from the PBX: for example I get incoming calls after seconds or I get no audio during calls. I thought it was something connected to DNS resolution so I put VoIP provider addresses inside /etc/hosts but
2007 Nov 26
1
OT: Best firmware for Linksys Router that is "SIP AWARE"
Hi, I am currently playing with DD-WRT and I like it. I am looking for something that is more "SIP Aware". Anyone know one those that are out there ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071126/eb28ce44/attachment.htm
2006 Dec 03
1
RTP Media Path
I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa (ATA -----> Asterisk ----> SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP