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I am wondering what is the best way to send the smallest files with the vm to emai l integration? I am not sure what order the three lines of the format command take, so I have just tried trial and error swapping. I think when set to "gsm" I get the smallest sizes. I can get my Windows Media player to play at least part of the file (get missing codec message from Realplayer), but get a

I'm having a bit of a problem here: I have a * box with a fritz isdn card (running capi 2.0 and chan_capi) and a x100p card for testing purposes. As a proof of concept, I wanted to be able to dial into the * using the isdn line, listen to a message, and enter a 3 digit extension number. If this happens, I wanted the * box to dial out using the x100p card, into our PBX (Nortel Meridian). If

you can always do a "restart when convenient" within asterisk, and it will do it's thing when all lines are clear.... -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, January 08, 2004 12:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Is there a way to reload a module from the

hi, i.m. newbie in asterisk. asterisk 1.0.3 is my current version. i like to store usernames and passwords in a sql database. i like to log failed authentification-passwords, to create a blacklist for securityreasons. i thingk a sql-database is a good way to log these actions. i don.t find debugging-options to output invalid login-passwords. Ok, i have made the following: debian is my OS.

Hi Everyone ;) I have posted something like this before but yeilded no solid help as of yet. I am new to * and havent even setup a box for it yet as to I have no clue what I should go ahead and buy before wasting a few $k. Im looking to setup * for my office with outbound calling only with some call agents, and also remote agents so they can work from home. At this time im not looking to

Hello all! I am trying to load sip.conf from mysql database. I have followed the instructions at <http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers>. Seems that the authentication (user & psw) works fine but I would like to get more information from mysql and I don't know how to retrieve it. Could anybody help me? Any idea about how to do it? Regards, Victor.

Hi, This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the "New York" time? What value I have to give to GMT offset value? Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

I am trying to cluster some binary data using k-means . As the regular "kmeans" available from stats package in R does'nt provide the option to change the distance method. I was wondering there is any package available to specify type of distance measure to be used in k means clustering in R. Especially distances like "Jaccard" which is good for binary data.

Might be a stupid question, but is there a default gateway set on the 7960? -----Original Message----- From: Paul Mahler [mailto:pmahler@signate.com] Sent: Thursday, December 18, 2003 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT? I have a 7960 running behind a firewall running NAT. From a telnet session to the 7960, I can't ping

Hi I have setup a Samba share in the Linux RedHat 8.0 server running samba 2.2.5 and created a user to have access to that share. So that any person attempting to access the share via a win2k machine will access it by providing user name and password of the share. I have tried almost everything ie, PC guest, smbpasswd etc. but still have no success. In the conf file I am using security = server

Hello, I suspect that r260758 by kparzysz at codeaurora.org broke build the windows bots which are using Visual Studio 2013 or lower. This doc here specifies that the minimum version required is VS-2013: http://llvm.org/docs/GettingStarted.html#host-c-toolchain-both-compiler-and-standard-library I do see that the bots running VS-2014 or higher are green. Can we either revert the said commit or

Hello, I'm wondering if someone may be able to help me, and do apologize if there is a simple and obvious solution for this. I am somewhat new to R, and have been searching for a simple solution for a couple of days. I am interested in finding a tool that allows me to plot a stacked bar plot. My data set is in the following format:

What is the proper method to install/configure an X100P FXO card? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031009/ce8a7803/attachment.htm

Hello all, I have no problems setting up trunk groups in general, but is there a way to set up a trunk group for outbound calls that includes channels on multiple servers? I might have missed something somewhere, but I couldn't find any reading about this topic. Thanks! Sean

On Wed, Sep 18, 2019 at 11:49 PM Mehdi AMINI <joker.eph at gmail.com> wrote: > > Hi, > > What happened since you committed the doc? Is this now graduated from a "proposal to start a project" to a project? I think it has now. > Is there any update to be made to the doc: http://llvm.org/docs//Proposals/LLVMLibC.html ? Not sure what kind of update you have in mind.

How can i configure the following scenario, - User 'A' dials into Asterisk, - Asterisk puts user 'A' on hold - Dials Out to User 'B' - Consults user B' if he wants to take the call (Press 1) or divert to voicemail (press 2) - Depending on the option chosen, either user A' call is bridged with the out call or transfered to voicemail. Thanks, Sharath Chandra

I am migrating from 64-bit RHEL 5 to 64-bit Fedora 12 and I'm getting the following error when I run ./configure in the directory R-2.10.1... checking for IceConnectionNumber in -lICE... yes checking X11/Intrinsic.h usability... no checking X11/Intrinsic.h presence... no checking for X11/Intrinsic.h... no configure: error: --with-x=yes (default) and X11 headers/libs are not available

Gentleman, In one of my slackware servers I put samba acting as a printer server connected to a Hewlett-Packard 2420D (D stands for duplex). In CUPS I'm using the 'raw' driver. So I let the clients to perform the driver stuff. On samba, I followed the procedures described on http://www.samba.org/samba/docs/man/Samba-HOWTO-Collection/classicalprinting.html#inst-rpc to make

Hello again, I would like to announce that we now have dedicated mailing lists for the llvm-libc project: 1. libc-dev - https://lists.llvm.org/cgi-bin/mailman/listinfo/libc-dev 2. libc-commits - https://lists.llvm.org/cgi-bin/mailman/listinfo/libc-commits A brief README.txt and LICENSE.txt have been added: https://github.com/llvm/llvm-project/tree/master/libc I will soon share or put out

hi, I have the following configuration: Grandstream --> NAT (Netgear RP614)-->Internet-->Asterisk(public IP) i can register fine and call ringing is working as good. The problem is = i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable my sip.conf file is as follows: