similar to: X100P "rings" randomly when "phone" line makes call

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

Hello all, I have a question. I have an asterisk box setup with an x100p card installed. I have about 3 VoIP phones connected to it. The x100p is connected to my pots line. I also have regular phones connected the the pots line as well like most normal people would. When I get an incomming call all the regular phones ring and my VoIP phones ring. This is good. My question is this. Say

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

Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms). I can run the same application from a GrandStream phone (on the same LAN) and hear little delay. What could possibly be wrong? If it were interrupt overload, I'd hear lots of cracks in my echo, right? I'm not hearing that. Besides, a telephony card is not involved.

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

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

With one IVR payment system, I noticed quite a difference in DTMF transmission between these two cards. The IVR missed nearly all digits from X100P, while receiving digits from TDM fine. Since neither card process or synthesize audio, what can the difference be? (This particular IVR has problem with some regular phone devices, too.) Yuan Liu

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...

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

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

The SM56 MODEM manual says it does. But when used with zaptel 1.2.12, nothing shows up. Yuan Liu

I'm greatly surprised when testing an Asterisk box with 802.11g. Here's the topology: VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension | FXO ___ PSTN extension When I call a VoIP extension on that box (from a VoIP extension), voice is good. But when this box tries to bridge the call with a

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

My SM56 (Motorola X100P clone) has echo as hight as 38%, according to fxotune -d. But when trying to take action, it fxotune simply says it can't. ./fxotune -i3 -vvvv Running with parameters: doset=0 docalibrate=1 dodump=0 startdev=1 stopdev=252 calibtype=2 waveformtype=-1 delaytosilence=0 silencegoodfor=18

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 =>

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

I have a * box running * version 1.0.3 with two X100P line cards in it and Cisco 7960 IP phones. Everything seems to work pretty well with the exeption that the system hangs up on phone calls for no apparent reason. It does this on both incoming and outgoing calls through the POTS line (currently only have one). The only thing in the asterisk console with maximum verbousity is " -- Hungup

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

This is rather bizarre: My TDM11 (one FXS) rings a $10 passive phone with REN of 1.0B, a cheap speaker phone of 0.3B, and a cordless phone with marked REN of 0.0B. But it couldn't properly ring this 27935GE3-B (FCC ID G9H2-7930) cordless phone rated at merely 0.1B. Rarely, the phone will crack out an occasional weak and abrupt beap, but never a normal ring. Otherwise Asterisk and TDM400P