similar to: X100P disconnecting without any reason

Hi All, This isn't really a question, but it's an issue I experienced that was driving me crazy for a few days, so I thought it might be good for the archives. Basically what was happening was everytime a particular customer called (long distance), the line would disconnect immediately after answering. I thought it might have been the phone, so I swapped the phone with another - still

Hi all, Which is the practical (from your experience) limit of the number of X100P cards installed in a single Asterisk box? Asterisk can work reliable with 6 X100P cards in the same box? Anyone know when the 4 ports FXO Digium card will be available on the market? Many thanks, Dan P.S. Please do not aswer with RTFG ...tried before without success...:-))

Hi, I have problem with X100P clone card.I can not force it to work under Asterisk 1.4.27.1 and DAHDI Version: 2.2.0.2. I looked over and over on configuration and could not see any mistakes. Here are relevant configuration files. /etc/dahdi/system.conf > fxsks=1 > echocanceller=mg2,1 > loadzone = hu > defaultzone = hu > dahdi_tool > OK Wildcard X100P

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 have two X100P cards in the same system. I can use both of them to initiate and/or receive PSTN calls. I want now to define separate context for each of them, in oder to route inbound calls to different extensions. This is what I have now in zapata.conf file: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes callwaiting=yes echocancel=yes

I am finding that with the combination above, the zap channel will hang up, though the call has hung up. I am make all changes in my zapata.conf file that I can find on the internet with nothing fixing this issue. I have added busydetect=yes, busycount=6, and made sure that the signalling is set for fxs_ks these setting worked fine with zaptel 1.2 but not with the newest zaptel drivers.

install asterisk with x100p clone; the problem is that call me and hangup but the interface zap not detect the hangup and the line open. the error is : The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf':

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, 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, 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 =>

Dear All, I am a newbie about asterisk. I have 1x X100P card 3x Sip phone I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I got no caller id, even my direct PSTN service operator. So at that moment I was using *1.0.9. than I changed to asterisk@home 1.3(1.0.9). I got same

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

Hello all, I have just installed a Wildcard X100P into an Asterisk box. I connected the line socket to the internal telephone system where I work. The card is identified to asterisk etc, however I am unable to recieve or make calls. When attempting to dial I get: Executing Dial("SIP/1106-ec8b", "Zap/1/644746") in new stack Called 1/644746 Zap/1-1 answered SIP/1106-ec8b Hungup

Just to make sure this isn't a typo in your original email... Is this example from your zapata.conf? Also, the extension you have shown are in extensions.conf not zapata.conf correct? Here is an example of a good zapata.conf.... [channels] language=en busydetect=yes faxdetect=both busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes useincomingcalleridonzaptransfer=yes

Hi All; Why busydetect=yes caused this autmatic hangup happens without any reson (while responding to entering the digits in the IVR) I do not know! And what is the solution I do not know. I used busydetect=yes and busycount=5 in zapata.conf to help me in hangup when detect the busy tone, but I faced the following problem: While I am calling to the Asterisk PBX, and during the response for the

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the

Hi All, When the Caller hangup at the voice menu, the wildcard X100P didn't disconnect the calls properly and it just keep looping at the voice menu and timeout and loop again, are there any methods can fix the problems? Please help! Thanks, Randal

Hi, I'm currently running asterisk 1.0.9-r1 and zaptel 1.0.9_p1-r1 (the current gentoo ~x86 versions), with the UK CallerID patches from http://www.lusyn.com/asterisk/patches.html applied. The Zap interface itself seems to work fairly well - although it's a little quiet, there is no echo. Unfortunately, there's also no CallerID. My zapata.conf is as follows: [channels]

I've configured an analog/single line off my mitel sx-2000 to a x100p card in my *, however when the remote caller drops the line I get a dialtone back from my pbx and the x100p doesn't detect the end of the call. Any way I can tell * to drop? I've tried call progress but it doesn't seem to work...

Hello All, I have this strange problem, I can dial out with my sip phone and it seems to work relatively well, but when I call in, the line just rings and rings, I get no indication in asterisk that it's detecting an incoming call. The strange thing is that in ztmonitor 1 -vv the rx volume goes from around 99% when the line is idle to 0% as soon as the line starts ringing. I'm located