similar to: Stress Test new system

Hi All, Have completely moved off the old ESI system, and things have been going pretty good with the new server. I have one issue, which has been reported by several of our customers. I've tested it, and it does indeed seem to be a problem. When the customer is asked to dial in the first three letters of the person they are trying to reach, they will be routed to the wrong extension.

All: Yesterday I discovered something interesting. I dialed 1800ANCESTRY from the asterisk system I am testing and got the number doesn't exist message. I then dialed the same number from our old system and it went through. I realized that the "Y" in ancestry made the number too long, and went back to my dialplan. How do I ignore numbers that are too long? Obviously,

All: RKG needs an asterisk consultant to help us track down issues we are having with our system. Mainly dropouts and dropped calls. If you have experience in troubleshooting these issues, please contact me at email attached to this messages. Regards, Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com --

All: My company has an existing ESI IVX E-class system with 45 phones. I can add one more card, to expand it another 6 phones, but it's $8000, and then the system will have to be replaced. I have the Asterisk server up and running, with 2 sip lines from the local phone service. (Thanks to you guys, it is working great!). I'm pretty sure this is the way the company will move, and

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

All: Still experimenting with the asterisk server for the company. My local phone company has given me two sip numbers to experiment with, say 444-456-1234 & 444-456-5678 Calling in and out works, and I've spread a couple of the phones out with some co-workers. My question is this: Do I have to define multiple sip lines in either the sip.conf or the extensions.conf? Now when I

All: Some of the people who dial into to our system will press the pound key when entering an extension for the directory key. When waitexten gets that, I get an error messages as, for example 123# doesn't match any extension. I was going to use ${EXTEN} to just use the first three numbers, but I'm not sure how to use this with WaitExten. so I have exten =>

All: I am looking to install another asterisk server in an office located in a different part of the country. I think I can configure the sip and extension conf files, so that the internal phones at the two locations can call each other. My question is this, how do I properly configure the sip file for a different provider at the new location? Can I use a different register statement for

"sip show channels" shows some info about active sip channels, the current codec included. What does it say? jg" jg, sip show channels reports the Format as being ulaw for 17 active calls. Holds - no Peer User/ANR Call ID Format Hold Last Message Expiry Peer xxxxxxxxxx kbrown xxxxxxxx (ulaw) No Rx:

Hello, Several people report that outgoing phone calls to our clients sound muffled, like they are talking underwater. Reported for both the Snom 870, and the polycom ip650. Incoming calls sound ok. Could this be a codec problem? My dialplan looks like: [general] port = 5060 bindaddr = 0.0.0.0 srvlookup = no tos_sip = cs7 tos_audio = ef registertimeout = 1 relaxdtmf = yes context =

All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an option I am over-looking on the grandstream, but if anyone can point me in the write direction, I would certainly appreciate the help.

All, I'm looking for sip phones that support something other than openvpn. There are a lot of vendors in China (mainly Alibaba) that sell l2tp VPN phones. Are there any American vendors that support l2tp? Thanks, -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com -- <http://www.rimmkaufman.com>

All: I've starting building an asterisk system for our company, which has about 60 users. I am new to asterisk, so thank you for your patience. I've stripped the sip.conf and the extensions.conf down to the bare minimum: Here is my extensions.conf file [globals] [general] autofallthrough=no [default] [fromprovider] exten => YYYYYYYYYY,1,Dial(SIP/151,20) [phones] exten =>

Am 19.08.2015 um 19:07 schrieb Steve Edwards: > Please don't top post. > > On Wed, 19 Aug 2015, James Cass wrote: > >> Steve, would you be willing to share that "quick bash script"? > > There's no magic in the script, but here it is, embarrassing myself: > > cp sample-call-file /tmp/ > chmod +x /tmp/sample-call-file >

Steve, would you be willing to share that "quick bash script"? James Cass <http://goog_987864563> jcass78 at gmail.com On Wed, Aug 19, 2015 at 12:11 PM, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Wed, 19 Aug 2015, Dominique Haeber wrote: > > Hi Barry Flanagan, >> >> Barry Flanagan <barryf-lists at flanagan.ie> schrieb am Mit, 19.

I'm using an SVM as I've seen a paper that reported extremely good results. I'm not having such luck. I'm also interested in ideas for other approaches to the problem that can also be applied to general problems (no assuming that we're looking for spirals). Here is my code: library(mlbench) library(e1071) raw <- mlbench.spirals(194, 2) spiral <-

Hi. I'm trying to plot a pair of intertwined spirals and an svm that separates them. I'm having some trouble. Here's what I tried. > library(mlbench) > library(e1071) Loading required package: class > raw <- mlbench.spirals(200,2) > spiral <- data.frame(class=as.factor(raw$classes), x=raw$x[,1], y=raw$x[,2]) > m <- svm(class~., data=spiral) > plot(m,

Hello: Does anyone know of a list of milliwatt test numbers for debugging echo? Specifically I am looking for a milliwatt test number in Canada, preferrably in a 416 or 905 NPA exchange....different carriers would also be nice....ie. Bell Canada, GT, Sprint (Now Rogers Telecom) I called Rogers NOC and asked them for the milliwatt test number....they didn't even know what it was....so I

Hi, some days ago we discused here the need for an analyzer for the 1000 Hz tone, as opposite application to Milliwatt. Here it is: Mwanalyze http://planinternet.net/download/voip/asterisk/app_mwanalyze.c It performs a Fourier analysis for a fixed frequency and tells the amplitude. The frequency is not limited to 1000 Hz, but can be passed as argument. The periode duration must be a mulitple