search for: peirce

...even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed the behaviour? I thought that when ChanSpy, MixMonitor, and the like are enabled on a channel it would be prevented from reinviting the audio to bypass asterisk. Thanks, Trevor Peirce

...with a Cause Code for Unassigned(1) and Congestion (42). Busy works fine though... Anyhow, they are seeing the RELEASE COMPLETE message with cause code 1, however the tech told me they expect a PROGRESS indicator with a value between 1 and 10. Any ideas on how to resolve this? Thanks, Trevor Peirce

...h voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists. Thanks, Trevor Peirce

...ler is transferred, call quality deteriorates. There is a pulsating noise that causes the caller and new callee to barely be able to hear each other. This happens every time and only on the 2nd call. The first call is always clear. Any suggestions are most welcome. Thanks in advance, Trevor Peirce

test only. good luck! james.zhu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/0fd2b827/attachment.htm

...to park a call and still be able to hear the parking lot stall number. Unfortunately, I have no idea where I saw that (google turned up little, couldn't find it on the list either). I'm using Sipura SPA-2000 adapters and it doesn't seem to work with today's CVS. Thanks, Trevor Peirce

...party receiving the transferred call still hears the double ring. Now, I'm not sure if this is design or bug. I think I saw mention of the _ doing something special but I'm not sure where and see no mention of it in the wiki under SetVar. Pointers would be appreciated. Thanks, Trevor Peirce

...kely to get a good answer! I'm interested in the provenance of the name "normal distribution" (for what I'd really prefer to call the "Gaussian" distribution). According to Wikipedia, "The name "normal distribution" was coined independently by Charles S. Peirce, Francis Galton and Wilhelm Lexis around 1875." So be it, if that was the case -- but I would like to know why they chose the name "normal": what did they intend to convey? As background: I'm reflecting a bit on the usage in statistics of "everyday language" as techin...

...terminal as such: Terminal <-> SPA-2000 <-> Public Internet <-> * <-> PRI I'm hoping someone will tell me they have done this successfully and rarely experience dropped calls. Though I'd like to hear from anyone who has tried and failed as well. Thanks, Trevor Peirce

Hello all, I would like to rigorously test whether observations in my dataset are outliers. I guess all the main tests in R (Grubbs) impose the assumption of normality. My data is surely not normal, so I would like to use something else. As far as I can tell from wikipedia, Peirce's criterion is just that. The data I am interested in testing is: 1) Continuous on the unit interval 2) Discrete 3) Ordinal on 0 6. If you need more specifics, (1) refers to the gini index of inequality, (2) refers to measures for the number of assasinations, strikes, etc in a country, (3) re...

Hello, Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and HPEC 9.00.003? In particular, with a hardware configuration similar to: Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) I have two fully independent systems

This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls

Hi, I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? Be waiting.thanks a lot Marlo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/16cc52cd/attachment.htm

Hello all, I'm going to tackle learning C this week, and start writing my first * add-on/contribution; assuming it's actually worthy of contributing once it's done.. I think I've chosen a hefty project for my first go round here... I'd like to get some feedback from everyone on a FindMe/FollowMe spec I've put together. Before you read on, let me say, I don't want

So I recently developed a collection of bash scripts to run QC programs on the computer my company sells to customers before they ship out. Originally they would run off live cds, but we just made the switch to Live USB sticks for more automation and the benefits of persistence. Right now each of these sticks has syslinux 3.71 installed, and boots up into a custom debian system to auto-run the qc

...of that nature, that will allow me to connect two existing calls together just as the Dial does. I'd appreciate any pointers to docs that I may have missed... I've been studying Asterisk for a few weeks and haven't been able to find a solution to this yet. Thanks in advance, Trevor Peirce

In my hunt to track down my echo issues, I tried disabling all echo cancellation, suppression, adaption, on my SPA-2000 (Advanced section of the config, under Line 1/2). Then calling from one local extension to another. (SPA-2000 Line1, to Line2 on the same device) I was pretty shocked with the results, the echo was HORRIBLE! I even tried 3 different analog phones. Now, once I turned the echo

Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick

...|1") in new stack -- Goto (trevsip,failedcall-CHANUNAVAIL,1) -- Executing Congestion("SIP/dc1-aa6a", "") in new stack == Spawn extension (trevsip, failedcall-CONGESTION, 1) exited non-zero on 'SIP/dc1-aa6a' Any hints would be appreciated... Thanks, Trevor Peirce

I am using Console/DSP for an intercom. I want to play my MP3 collection over it when no one is using it, like when they do in the supermarket. Can anyone help me with this. Any suggestions will be appreciated. -- Christopher Dobbs