similar to: X100P knock-off price jump

I'm looking to get an x100p off ebay and am not particularly familar with the "life cycle" of the card.. An "Authentic X100P" listing has a buy it now of $29.95 and says There are 3 types of cards Asterisk would recognize: *Screenshots from the official, original driver install Cheap "OEM X100P","Clones", "Compatibles", Knock-Offs

Do they fall under FCC certification if they're built to the same specifications as the ones from Digium? If I build my own T100Ps from the schematics and board layouts that are available, are they legal to plug into the PSTN?

Hi, I am the "IT guy" at a small startup based in UK. At the moment we have 3 analogue (PSTN) lines and we will be adding another 2 or 3 soon. Later on we should be changing to ISDN30. One of the partners mentioned getting an analogue PBX now, and when we move to ISDN, then get a digital PBX. I though of Asterisk. I have seen the website in the past and I know that it can do the job

So I took a chance with an X100P knock-off on eBay. I'm running Asterisk + FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel 2.6.16.16. Everything has been fine up until now. I compile the 1.2.5 Zaptel drivers without a problem, get the udev configuration in, modprobe zaptel, and finally modprobe wcfxo. At this point, I get the message: ZT_CHANCONFIG failed on channel

Digium X100P / new cards are is available on ebay for $43. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 <http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 > Hope this helps to who want to play with X100P! Are these being sold by Digium ? I don't know ?? - SamW -------------- next part -------------- An HTML

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 am starting to dip my toes into the asterisk world, and to that end I've scavenged an old PC (this is a home project, and I have basically $0 to spend on it), and installed FC1. I've purchased a clone X100P (or at least I was told it was), and I am trying to get it to work. However, when I try and load the wcfxo module, I get the following errors... [root@loudhoward root]# insmod

If Asterisk is configured as a simple answering machine replacement with the X100P connected to PSTN line. No FXS ports in the Asterisk machine. Standard phones are connect in parallel with the X100P like you would a regular answering machine. Can Asterisk detect that a phone has been picked up and cancel the outgoing message and/or voice recording? What about if the phones are connected to

I am attempting to use the Knock.pm from http://www.shorewall.net/ManualChains.html I am not having much luck making the DNAT- knock work for some reason. Anyone else using this on 4.4.4 that can verify if this still works as documented? Thanks ------------------------------------------------------------------------------ Return on Information: Google Enterprise Search pays you back Get the

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

Ryan, from the console what does "zap show channel 1" or 2/3/4 in your case say. I have X100P's and I seem to be having similar sounding problems. I noticed that the above command shows the channel to be off-hook at all times when a phone line is plugged in. I don't know why or if it is a bug in the application reporting the status. dbc. Ryan Courtnage wrote: > On July 8,

We have been pretty much able to solve our echo problems, except for the primary mode in which we desire to operate our system. See system diagram at bottom. Prior to making adjustments to cancel echos (all echocancel=no): Call Type Result (Before) --------- --------------- CP <- LEC PRI * TDMoE * FXO -> AP

Hello All, I installed Asterisk@Home .5 last night. I was able to configure some extensions for the house and they work fine. I just can't make inbound and/or outbound calls. The Flash Operator Panel shows four external icons and my new extensions. I have four X100P and two Broadvoice sip accounts. Thanks, David -- David Shaw <asterisk@ke6upi.com>

Hi, I am quite new to asterisk so I am not sure what is needed to figure out this problem. If more information is needed and not provided I will gladly provide it. I have a very basic asterisk setup. 1 x100p card and a grandstream handytone 286. I can make calls fine to most phone numbers from the handytone device the trouble seems to come when I dial this number 591-1079. It puts me through to

Hi I thought this might be of general interest. Recently I purchased an X100P from a Digium reseller in the UK. Very pleased with the card; works perfectly. My friend (known for his deep pockets and short arms) purchased an X100P card from Ebay. He's had no end of problems with line noise, dropped called etc so I thought I would compare the two cards. Plus mine was delivered within two

Can someone please tell me if this is how call waiting should work? ZAP/1 and ZAP/2 are X100P on phone lines ZAP/3,4,5 have phones conected. If a call is in progress between ZAP/2 and ZAP/3 and ZAP/2 gets a callwaiting signal which ZAP/3 hears the Dial command has SIP/2002&ZAP/3&ZAP/4&ZAP/5 If ZAP/4 is off hook, I hear the call waiting signal The CLI says it is ring the 4

Hi guys, I've made a patch adding LocalPreCommand to ssh_config. It mimics behaviour of LocalCommand, but is executed right before the connection is opened. This makes possible e.g. to integrate ssh with port knocking. It also removes "-oPermitLocalCommand=no" from scp allowing the same functionality to be used for file transfers. Applies cleanly on vanilla OpenSSH 6.7p1.

Hi all. I`ve installed a X100P on my box and is working well with incoming and outgoing calls as a trunk with one PTSN line. I want to connect the X100P to my Panasonic 616 EASA-PHONE as an internal extension to permit to users to make calls to SIP devices from analog phones, the problem is when I dial the ext number where the X100P is connected I get busy tone. What config I need to change to

Hello my brothers and sisters, Is "X100P" card suitable to VoIP? and if "yes", am i need to only "X100P" and "Asterisk" Package? or i need also to other cards or packages? and if "X100P" card not suitable to VoIP, please recommend a another card, (please take in your account that i would like to connect standard analog line to the card

Hello, Can the wildcard x100P be bought of the shelf in stores - say in Singapore. Thanks Varun