similar to: re: where can i find all .C files

this is not just playback recorded voice. this is let asterisk say chinese. how to do this. there have any ideas? -- Jeffery iaxtel Num: 1-700-576-1311 fwdnet Num: 728150

Or you can use authenticate() and have it take its 'passwords' form a text file on your machine. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > trixter aka Bret McDanel > Sent: Sunday, January 29, 2006 5:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion

Hello, I am wondering about the ability of a server that is simply passing G729 through it to have the ability to record the calls. I know for voicemail, meetme, and things like that to work, a G729 license must be installed on the machine since there is transcoding going on. Is this also true for recording of calls? Will I require licensing for each recorded call? Will the server see a

Hi there! I just try to play with die wake-up function described in http://www.voip-info.org/wiki-Asterisk+tips+wake-up Everything looks fine but there seem to be missing some soundfiles like "wakeup-menu". Where can I get these files in order to make this feature usable? Regards Julian Pawlowski

I am wondering how the g729 licenses are done during calls. If I have N licenses for g729, and N are in use and an additional call comes in that requests N+1 to be in use, how does asterisk handle that call? Does it dump it? Does it negotiate another codec automagically? Basically what happens to that call, obviously it wont (shouldnt) let you use more licenses than you have available, but

Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Cheers, Richard. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060109/26c4a63c/attachment.htm

Hi Tejas, Thank you for your interest in Letor project in Xapian. We would definitely like to consider Letor for this year's GSoC project. What I would suggest you is, start playing with the code and get acquainted with it. The latest code can be obtained from http://trac.xapian.org/wiki/ GSoC2012/LTR Regards, Parth. On Wed, Jan 22, 2014 at 10:14 PM, Tejas Nikumbh <tejasnikumbh at

hello all, I m joined this asterisk group few months back. Actually i have installed asterisk on my PC and using X100P PSTN interface card to make this PC work as a single line VoIP gateway. Now i have one problem which is as follows: As i told i m using X100P card. I am getting good quality of speech from it whenever i m making call or receiving call through it. Now I have

HI: what is command on console to know how many calls are sending in the same time? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Regarding my earlier post about labels and the 'n' priority: The TFOT book covers the use of these. See the box on page 81 entitled "Unnumbered Priorities." http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip -MC

If i get a 8XX number, my provider told me that they will send all the calls he gets. But due to bandwidth and asterisk capacitiy in this particular installation, the system is only able to handle 27 incoming calls. How in my dialplan do I regulate, sending a busy signal, when my system hits 27 incoming calls? thanks, -- ------------------------------------------- Erick Perez

Hey guys, I had introduced myself earlier on IRC. I talked to Parth and had a brief chat with Olly, but just to re-iterate I'm Tejas Nikumbh, and I'm interested in contributing to Xapian for GSoC this year. I'm specifically interested in letor and weighing schemes projects. I've been able to build xapian on my machine without any errors via the Guidelines on the Hacking

I want to create a VoIP solution to allow many members of a closed community to talk to each other (one on one) via soft phones. In many ways, what I want is not unlike Skype, except that it would allow for relative anonymity and be open only to a select group. The system should support as many as 500 simultaneous users located on different continents. Is Asterisk an appropriate tool to use in

After experiencing these symptoms through many of the release candidates, we've been at v1.0.0 now for a few weeks, and they're still there. Perhaps a config issue with my setup (mbox locks?). It's really too disrupting to continue using this way. I've included symptoms, client info, mta info, log entries and config file. What other information can I provide? Please

Hello, Thought people might be interested in this. I want my voicemails emailed to a person and not stored on my asterisk server. However, I want them to have a sequential number. I found that if I set the option delete=1 in my voicemail.conf file for the mailbox, then the numbering would keep being restarted. I wrote this shell script to fool Asterisk into numbering my voicemails sequentially

I have searched and found a couple examples on how to put the app_cepstral.c into Asterisk but it isn't working. I obviously am not understanding the examples that I have found. "Copy the app_cepstral.c file to your asterisk source tree (apps folder) . You'll also need to add a lines like these to the Makefile in that directory: APPS+=app_cepstral.so" I added it into

Hang on.... there's a non commercial G729 codec that will work with Asterisk? Can someone point me to where I can find it? Thanks, Doug. -----Original Message----- From: Francesco Peeters (Asterisk) [mailto:francesco@fampeeters.com] Sent: Sun 1/22/2006 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion

Hi All I have some grandstream phones registered to my asterisk and all internal, external, voicemail services etc are working very well. I am not sure that it is a problem more so an annoyance. If someone dials my extension number or external DDI while I am already in a call rather than skipping to the next priority in the dial plan for example voicemail the line continues to ring and while in

Hello, I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or some other system to deliver to an e-mail address(s). I'm not sure if I need Asterisk to any of the call control or not. I'd also like to setup a print queue and have outbound

Greetings Richard, To add to the above information, we are following the understated steps: 1. We have 2 BL460c (Gen8) with Fedora 21 installed. Both the machines(BL1 & BL2) have been updated with virt-v2v packages 2. On BL1, we execute the following commands: 1. virt-p2v-make-kickstart fedora 2. livecd-creator p2v.ks 3. After creating the