similar to: PSTN line tests

DSL works by using the frequencies above 4k that were unused in POTS loops of yesterday. Load Coils, Bridge Taps, and DC taps are all devices added to lines to increase their reach and stability, unfortunately, they are DEADLY for DSL. Other problems can effect DSL service, and cause it to be 'flaky'. 1 Temperature, in Florida the large black cables are constantly beaten down by the sun,

I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone

BUG/SYMPTOMS: 1.Under certain circumstances, octoBRI (and most likely quadBRI) ISDN cards (Junghanns/CologneChip) severely distort certain ISDN payload. 2.Although these claims relate to the bristuff patch, the problem might not be limited to bristuff and in fact be rather asterisk/zaptel/libpri related. 3.Signal distortion is limited to the use of the CVS version of the Bristuff patch for

I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this

I'm not sure where to check next, so I'm reaching out to those that know this stuff better than I. I've got Asterisk up and running, but I've still got an occasional audio issue. Once in a while (maybe 1 out of every 20-30 calls), the audio becomes heavily distorted, but only on the local side. The party on the other end says the audio is fine. We can hear them, although

Hi All, Last week I started hearing a huge amount of local end echo on incomming calls. I am using a Sipura SPA-3000 as my FXO connected to an SBC POTS line. Echo cancellation is enabled in the SPA firmware. As a test I switched to a Digium X100 card the still lives in my server but the echo was about the same. I have both Polycom IP600 and SNOM 200 phone, which both hear the echo. I'm

On Sun, Jun 24, 2001 at 01:51:05AM -0700, Dave Hayes wrote: > How are people doing gain control, out of curiosity? > > I know boxes exist (anyone have names and mfrs?) that ride the gain > for big radio stations, ensuring that there is no distortion and > raising the volume of songs recorded at a lower volume. I'd probably > buy one if I knew what to buy. This is a bit

Brent- There is no good way to say what changing the hardware and PSTN hookup will probably do for the echo problems. I'm not sure if you mentioned (lost in the past history of your post now) what sort of hardware you're using for PSTN connection now- TDMs, X100s, ATA's, etc- but that could also be a potential cause. I've heard tell of aftermarket X100s and certain ATA's

-----Forwarded Message----- From: IEG <dennis.andring@gmail.com> To: derek@kfuq.net Subject: Re: [Fwd: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM] Date: Tue, 13 Sep 2005 03:04:42 -0700 The answer is a multiplexed terminal node controller (TNC) This was the very thought behind "trunked" communications around 800mhz. Gee ... there are a bunch of cell phone

I'm looking for a function in R similar to t.test() which was generously pointed out to me yesterday, but which can be used for normally distributed data. To recap yesterday: > x <- scan() 1: 62 52 68 23 34 45 27 42 83 56 40 12: Read 11 items > alpha<- .05 > t.test(x) One Sample t-test data: x t = 8.8696, df = 10, p-value = 4.717e-06 alternative hypothesis: true

Hi all, Currently trying to get asterisk to dial out with an Internet Line Jack card, however, it does not use the pots line, only on the line it dials out of. This is similar to the previous thread/posting "Asterisk won't answer pstn ring", but I didn't find any follow up to get it working. My asterisk setup is like this: iptelephony:/etc/asterisk# cat phone.conf | grep -v

Hello I'm using a Linksys 3102 as VoIP gateway to connect a POTS line to a PBX. I also have an IP phone in a remote network across the Net. The PBX + gateway, and the phone are both behind a NAT router. I was wondering: 1. When a customer calls us through the POTS line and I pick up the call with the remote IP phone, do RTP packets go directly from the VoIP gateway to the IP phone, or

My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: --------- In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If

Dear All: I am trying to shade the area between the two lines; *line 1* and *line 2*. You can use this code as an example. x100<-c(-1,1,2,3,4,5,6,3) y100<-c(4,5,3,1,4,4,2,-1) plot(x100,y100) *##### line1* abline(a=-(Beta0-1)/Beta[1,2], b=-Beta[1,1]/Beta[1,2], lwd = 3, col="skyblue", lty=3) ##### lty=3, *##### line 2* abline(a=-(Beta0+1)/Beta[1,2],

Not sure if anyone experienced the same - or if anyone ever connected a POTS phone to the "Phone" jack on an X100P card. The POTS phone rings normally when the FXO receives a call. The POTS phone can also make outgoing calls when FXO is not holding the line. This is desired. But if a call connected to the POTS phone lasts longer than a couple of minutes, Asterisk would receive

I'm about to purchase an adaptor for a POTS data modem and was looking at the Sipura line of adaptors (SPA-1000, SPA-1001, SPA-2000, SPA-3000). Do these work well? Anyone have a suggestion on which model of the Sipura I should get? Does one work better with * than the others? Are there other adaptors that work better that I should get? Thanks, -Dave -------------- next part

Does anyone have any idea what would cause distorted audio but ONLY for DTMF tones coming in over our analog lines. (The analog interfaces are X100P's). I have carefully adjusted the gains in the zapata.conf using a local test line after trying various settings with no gain or just random gain settings. RelaxDTMF has no effect. I set up a monitor command in my dial plan to capture

Hi! > > > Since it is a PCIe card, it does not have the ability to host hardware > > > devices for networking, storage and console. We provide these devices > > > on X100 coprocessors thus enabling a self-bootable equivalent environment > > > for applications. A key benefit of our solution is that it leverages > > > the standard virtio framework for

Ok maybe it's just my imagination, but it looks like the one-line "duh" fix I committed from Andrej Vakrcka <ander@cauldron.sk> this week DRAMATICALLY reduces distortion in encoded files. Take a look and compair. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected to the one FXO and two phone sets connected to the FXS port. Extension 6003 is asigned to one fxs and 6004 is asigned to another fxs, the two extensions can call each other, they can both make/receive PSTN call, but they can't make PSTN call simultaneously. Is it