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to a fax (modulation and demodulation) over VoIP. Try again, the fact that you succeeded twice was pure luck, and as far as I understand that didn't even work out. Switch back to TDM. Your dial up modems want that magic thing called timing and no jitter that only TDM will give you. On Mon, Aug 29, 2011 at 2:56 PM, Aaron Krohn <akrohn at ewebforce.net> wrote: > We have an Ascend Max

Hello, I am currently trying to configure an Asterisk (current 1.4) box with a TDM410P and 2 FXS as well as 2FXO modules. An analog modem is directly connected to the TDM (port 4) which itself is directly connected to the PSTN (port 1). [V.34 Modem]-----[TDM on Ast. 1.4 TDM]---PSTN When I try to establish an analog modem connection with this setup, the modem is unable to connect to the remote

On 15-06-15 08:48 PM, Matt Darnell wrote: > In the past we have used Adtran Atlas 550's to break out FXS ports for > devices like modems. The great thing about the 550 is that internally > it is all TDM so there is absolutely zero latency. > > We are able to use ATA's for faxes and analog phones but devices that > use modems, they fail 99.99% of the time when using an

Hello, I'm running into a few situations on lossy network links where a SIP jitter buffer w/ some PLC would be helpful. My main TDM gateways are running 1.2 (which is solid, stable, reliable and very very very well behaved when you know it's limitations), but I'm considering upgrading them before the end of the year to 1.4. Two of the main reasons that I would do this are Variable

Around 1978, when I was consulting to a multinational company in the business of agriculture, I witnessed this configuration in their communications center in NYC: A paper tape punch attached to a teletype machine was busily punching out a tape that was being spewed into a wastebasket. Somehow, running behind it by several feet of tape, was a paper tape reader on another teletype drawing

In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but devices that use modems, they fail 99.99% of the time when using an ATA. We tried to migrate to TA908 devices; they have FXS ports built

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

Hello, we would like to hook up analog modems behind an Asterisk server, and we're very interested in the experiences that others have made when attempting that. We assume that there are no inherent problems with modems in respect to the Asterisk software, but it appears that the FXO/FXS hardware restricts this kind of a setup to lower data transmission rates, is this correct? Currently,

Hello I'm looking at my options to build a compact, silent, headless Asterisk server to handle one or two FXO ports. Out of curiosity, I got one of those babies on eBay for 20E: http://silicon-verl.de/home/flo/software/netstation-8364/ Before I spend time on this, can someone tell me if... 1. it's easy to install Linux on those things (am willing to add a CF-to-IDE adapter to use a

Hi all, I like to have a dial in modem on my toll free number so that when I or my employees travel, they can always get in for net access to read email if no better method is available. Right now, my Panasonic KX-TD1232D PBX receives the call on a POTS line and routes it to an analog modem. The speeds achieved are around 24 kbps due to the digitization of the KX-TD1232D introducing

Hi All, I have a question regarding the nominal jitter buffer configuration: The call was setup as G.729A (annexb=no), ptime=20ms and nominal jitter buffer size = 50ms, and round trip delay is 200ms, the TDM side will experience intermittent one way voice during the call, but IP side can always heard the voice from TDM side. My question is, should this possible caused by the nominal jitter

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Dear all Please help me with analysis of some periodic data. I have an output from measurement each minute and this output is modulated by rotation of the equipment (approx 6.5 min/revolution). I can easily spot this frequency from spectrum(mydata, some suitable span) However from other analysis I suspect there is a longer term oscilation (about 70-80 min) I am not able to find it from

I just needed to test a dialup modem connection (don't ask) and I had a modem connected to a TDM card (FXS port) which then dialled out via a E1 PRI on a TE4xxp card. See my log below: atdt0198xxxxxx CONNECT 36000 V42bis ** Dial IP ** Username: xxxx Password: Entering PPP Session. IP address is xxx.xxx.xxx.xxx MTU is 1500. OK ATH0 OK ATDT0198xxxxxx CONNECT 31200 V42bis ** Dial

Thanks Brian, and thanks again for the included definitions <grin> - that helped too. Your comments are really helping clear many questions. I suppose our intensions are to become an IXC. So if my local carrier is sporting old technology, they'll provide TDM services. So if I understood you correctly, the "in-band signaling" is typically SS7, and the alternative is typically

Hello all, I've posted on this problem before. Well here goes again. I have an intel mobo with a p4 2.4ghz proc, 1GB Ram. It has built-in ethernet and vga and 6 pci slots. I dreamed of making this my household communications server: internet router, firewall, vpn and asterisk. Everything works except the TDM fxs card. Well it works for a little while and it dies: no dialtone, no

Steve Underwood, Would you mind summarizing where/how T.38 functions, and maybe how it compares to the analog fax environment for the asterisk-users arhives? Seems to be some misunderstanding, and a lot of interest in handling faxes in various forms via asterisk. If some these approaches were summarized in one posting, a lot of us could reference it to remind us of limitations, current state,

Our setup is: 9.6k Modem <-analog-> Mitel SX-200 <-(pri)-> Asterisk <-(pri) -> Telco The modem works fine with the Mitel directly connected to the Telco, but once we add Asterisk in between connections start failing. I suspect the issue is caused by the echo canceller, since I believe the issue appear about the time we turned echo cancellation on (for the IAX users). We

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

Hi: I' have one TDM400P and need configure my linux box like modems. It's posible give my users with PC/modem/windows/diul up line access to internet with my linux/asterisk/tcm400p?? Thanks in advance Edgardo