similar to: Asterisk at Home ...

No Dice so far, anyone now how to add anIAX trunk? What are the settings exactly? I have added everything but I do not know what are the registration strings? Jonathan

Hi, is there a way to take a data frame with 100+ columns and large data set to do efficient exploratory analysis in R with pairs? I find using pairs on the whole matrix is slow and the resulting matrix is tiny. Also the variable of interest for me is a binary var Y or N . Is there an efficient way to graphically view many variable relationships that does not look teeny ? I could do

Hello List! I have a tiny-teeny problem with dovecot + sieve: it seems that the LDA doesn't run sieve, and thus doesn't filter my emails. Here's the sieve configuration: plugin { # Used by both the Sieve plugin and the ManageSieve protocol sieve=/var/local/vmail/%n/dovecot.sieve sieve_dir=/var/local/vmail/%n/sieve sieve_extensions = +notify +imapflags } The

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,

hello is there a gui that would allow me to configure everything from phones, to extentions, to voice mail to basicly everything that asterisk can do? I did go to www.voip-info.org and none of the guis I saw there do the trick and the ones that come close aren't downloadable just wanted to see status on this thanks hank ---------------------------------------- My Inbox is protected by

On 1/20/20 4:26 AM, Martin Maechler wrote: > Coming late here -- after enjoying a proper weekend ;-) -- > I have been agreeing (with Spencer, IIUC) on this for a long > time (~ 3 yrs, or more?), namely that I've come to see it as a > "design bug" that rpois() {and similar} must return return typeof() "integer". > > More strongly, I'm actually pretty

Please bear with me as I know I have included a lot of detail. Description Redhat AS 3 Kernel 2.4.21-47.0.1.ELsmp eth0 HWaddr 00:07:E9:11:30:76 inet addr:192.168.1.3 Bcast:192.168.1.255 Mask:255.255.255.0 eth0:1 HWaddr 00:07:E9:11:30:76 inet addr:192.168.3.1 Bcast:192.168.3.255 Mask:255.255.255.0 This interface is up but with no IP address eth1

I have started to receive a lot of positive response for the Asterisk@Home project. For those of you unfamiliar with this project the goal of Asterisk@Home is to make a full featured version of Asterisk very easy to install. We have created a 1 step .iso that installs RHEL (RedHat Enterprise Linux) and Asterisk. It includes a web GUI that allows easy editing of the Asterisk Config files.

I'm evaluating CentOS 5.4 for our company and one of our requirements is that it must also run on laptops. I've managed to get everything working so far except for wireless networks. The problem here seems to be that CentOS fails to provide a knetworkmanager package, and we're using KDE3 for various reasons. Thus, we're limited to Networkmanager-gnome which works but fails to

I have an Asterisk setup with AMP installed. I have phone extensions from 7000 to 7010. I experience long delays when dialing a 9 digit number as opposed to a 10-digit number. How do you get around not having to press the # key to speed up the dialing process? For any length phone number for that matter-- like dialing another extension. If I dial 7005, I'll have to wait a while.. but

Ugh, sounds like competing priorities. * maintain type consistency * minimize storage (= current version, since 3.0.0) * maximize utility for large lambda (= proposed change) * keep user interface, and code, simple (e.g., it would be easy enough to add a switch that provided user control of int vs double return value) * backward compatibility On 2020-01-20 12:33 p.m., Martin Maechler

I want to set up a dial rule like this 9304752#w9#w+NXXXXXXXXX The point of this is. It will dial into a pbx with the account number 9304752, wait a second, dial 9 to get an outside line, wait a second for the outside line, and then dial the number to be called. When ever I save this in amp anything after the first # disappears. When I try to call out it doesn't do the adding of the

Hi, I can't figure out how to change User Login and Password for AMP. By default it is user:admin and password:maint. Anybody knows how to do it? Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/c8ef57b2/attachment.htm

I'm new to this and have tried to find the answer in the discussions and docs but to no avail. I even read the posting saying the password issue has already been discussed. So, at he risk of being exiled, here goes. Question 1: I've installed 0.7 and can log into the asterisk server from windows by typing http://192.168.1.11 I can log in with wwwadmin and the password I set myself

Hi I installed *@home 0.6 to play with the system, and learn. I tried AMP but didn't like it and so set forth into the conf files manually. I have it all set up how I want, all my extensions work etc. Reading this list and playing in the wiki and google, I get the impression *@home is great for learning, but has a few limitiations. I would like to upgrade my box to the latest stable cvs

hi all, can anyone emailme the .conf of asterisk at home, i cant download the full size tar or iso because of a network problem that fu*** every big file download.... and i just wanna learn not change my distro bye and thanks! -- Luis Diaz - Un obsesivo con proyectos! :oP

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

Hello every one. This is a question done by me, not yet answered. Please, help. I: 1. Run install-pdf from linux to support faxes on my asterisk. 2. Made the configurations throuhg AMP in a. Setup->Inbound Routing->(the only route I have)->fax extension->System b. Setup->Inbound Routing->(the only route I have)->fax email->(my email) c.

I installed oh323 and everything seemed to go smoothly (compile & everything upto calling through using oh323). I must admit, there is some behavior that's doesn't seem right but generally, I'm able to dial-out of any oh323 device whether to an extension or to a trunk. Audio is sometimes muted when dialing out until the extension or dialed number answers. Sound quality is good

Folks, I recently obtained a Western Electric multi-line phone and am seeking help with getting this beast working with *. The interesting stuff in my * implementation consists of a T100P card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port FXS modules and a quad-port FXO. The TA750 is wired to a 24-port Cat 5 patch panel via a 25-pair Amp cable. The phone is a model