similar to: asterisk install in a home with regular phones and a x100p

Hello, After much searching of voip-info.org and google, I'm finally giving in and asking the list. The setup I have is this:- Single X100P card in a Debian system Inbound/Outbound POTS line connects to the X100P Sipura 2000 and Budgetone 100 on the LAN 1 Cordless and one conventional phone connected to the sipura Account on Stanaphone.com for eitherbound SIP calls. (I have other SIP

I have an X100P used, at present, largely for outgoing calls. It shares the single incoming POTS line with a number of analog phones. Is it possible to talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd like to use only the SIP phone in my office, but let the analog phones continue to work in the rest of the house (until I can afford FXS cards anyway..) I can force

I apoligize in advance for this newbie question on what I perceive as a mostly advanced level list... I did some searching, but would like some of your expert opinions. I'm building an asterisk server to be used in the home, both to learn, and as proof of concept of applying this solution in a home. To keep costs down, I'm considering one X100P ($25 ebay clone) card to connect the

Hi.. I've got an asterisk setup with an X100P card installed.. I'm noticing that upon hangup, it takes a good 3 to 5 seconds before asterisk realizes the line has been hung up and drops the call.. this causes my SIP phone to continue ringing, and occassional phantom voice mail messages to be left.. I'm located in good old standard North America, with a regular Verizon residential

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

Hi All... I have installed a single X100P card in my PC and am playing with Asterisk. The wire I plugged into the X100P has two POTS lines on it, wired on the RJ45 in the normal way. I am getting odd behavior. It seems when I dial out that the X100P dials both lines at the same time. I have two questions. First, I see that the X100P is only a single channel. Does this mean that I can

Noob here, my apologies in advance. Recently my employer decide to stop paying for my home POTS line, so I ordered packet8 for a home line instead of another POTS line, since I really dislike my local phone company, and the POTS line without any long distance would cost more than the $20 to packet8 with unlimited US calling. Anyway, this whole thing got me started thinking about VOIP a lot

So I've been on the Asterisk list for a long time. Mostly looking to set it up as a cheap alternative to a commercial MeetMe bridge, as a for-fun project. I keep noticing the Wildcard FXO X100P sorta seems like a Winmodem. Winmodems are basically a sound card chip tied to a pots port, where the Winders driver takes on the MO/DEM functions in software. I've always despised them as a data

I finally got my X100P working and now I have a question. I have several Skutch phone line simulators. My X100P works as expected with both a POTS line and an analog PBX port, but when I use a phone line simulator it doesn't answer the line. The phone line simulator doesn't power the line until the phone set goes offhook. Asterisk shows the RED alarm and then the alarm clearing but never

I know the X100P cards are not supported by Digium any more, but for home office use, are they still acceptable? I have two POTS lines, one residential and one business line comming into the house. I'd like to get both into my * server and $15 total compared to > $100 for the newer TDMxxx card sure is desirable. Having said that, will the sound quality, functionality, stability, etc. be as

Greetings, fellow Asterisk users! I'm having problems with Asterisk and Digium's X100P (using kewlstart) in Finland - Asterisk doesn't detect POTS hangup. I've modified indications.conf according to the Finnish specifications (well, I think I got the specs correctly :-), but still no go.. Has anybody used * in Finland? Does your * detect POTS hangup (which plays the

Thanks for the information Lee ! Still, something is still strange to me, since this two Panasonic Fax machines are receiving at least 20 multi-paged faxes a day (they are in same office as both Asterisk boxes, and me :) ). Beside that, POTS lines in those two faxes are from same PSTN switch as line in X100P in one of Asterisk boxes, and the ISDN PRI lines in other Asterisk box is from same PSTN

Hello, I'm looking for a dirt cheap solution for SOHO use to handle at most a couple of POTS lines, and I notice that X10?P cards go for $15 on eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma. I have a couple of questions about those cheap FXO cards: 1. Are they all glorified softmodems, ie. none has an on-board CPU or DSP and outsources all processing to the

OK, I've spent way more time than I wanted to on getting an x100p clone to work in Australia. I'm happy to consider other (more functional) options. Does anyone have an opinion on both the Sipura 3000 and other Digium cards (like the TDM400P)? I need something that works with no much fuzz. I know the Sipura 3000 is cheaper the the TDM400P card. All I need is to channel my POTS line

Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up

Hi All... Is it possible to cause asterisk to answer some distinctive ring patterns but not others? I have a POTS line plugged into an X100P and I would like * to ignore one ring cadence that is answered by a fax machine. How would I do that? Thanks...

Hi I've got a cheap card from x100p.com for my pots line. I haven't found a definate answer if it is possible to get danish(DTMF without signaling before it). I have had a look at bug #9 but that is written longtime ago. I am running zaptel 1.4.7 and asterisk 1.4.14 BRIstuffed 0.4.0-test4 both from the xorcom debian packages. I believe they should contain some kind functionality for DTMF

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

I have 2 X100P card and configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt I changed the area codes to match mine. When I try to dial out I get app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' A zap show channels gives me this Chan Extension Context Language MusicOnHold 1

Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k is installed I have a x100p card and it doesn't detect a hangup from the calling party when going in voicemail(). My PSTN provider is sending open loop disconnect (voltage decrease for a given moment of time). Actually Progress Detection is HIGHLY EXPERIMENTAL so it should not be required to fix this problem. I wonder if disconnect supervision is the