similar to: Pickup zap channel already in use?

Hello, I just bought a X100P from digitnetworks. It is supposed to be a FXO card, but there are 2 rj-11 plug on the card. One is labelled "phone" and the other "pstn". When i plug the "pstn" on the wall and the "phone" on my analog phone, everything (incoming and outgoing calls) works like before (without asterisk). AFAIU, i should have an FXS card in my

Hello, In my quest to figure out the source of the random echo on our shiny new asterisk install, I have been using ztmonitor on the TDM400p channels for the good part of today. I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still

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

Been trying to get Passenger installed and setup to run a rails app. Everytime I goto eaa.mysite.com in Firefox the web browser goes out to the Internet instead of serving up the website. Why does this occur? Below is part of the configuration for apache2.conf: ======================================================= # Setup a virtual host for the rails apps to run in. <VirtualHost *:80>

Many thanks in advance for anyone that can offer help on the following questions: Asterisk Box Using Asterisk@Home build and updated Asterisk to v2.1 P4, 400 Mhz, 384Mb RAM, 40Gb HD 4 OEM X100P Cards Phones Grandstream GXP-2000 2 * Grandstream BT-100 HandyTone 486 Sipura SPA-3000 Questions 1) When someone calls in to one of the FXO lines, there is a 3-4 second delay before the configured

I've read and experienced the echo problems with the X100P. Is Digium going to fix the problem or refund our money? I want to see this work because myself and other small companies out there use analog lines. I would trade up to T1 but that requires me to have at least 9 lines. If I did trade up, do the T1 cards work perfectly with no echo at all? I get echo with my directly connected

Hello! I've looked at the reference examples they are all for SIP. I have two X100p and a TDM400P. Can someone send me a working example so I can receive calls and make them. I'm stuck at first base. [I'm using standard phones - not SIP] Help please! Thanks, Bill Flood

Hi all, Really excited to be using Asterisk and learning about VOIP and PBX's. I'm a complete beginner at telephony but have built and installed Asterisk 1.4.5 and read several of the Asterisk books online and have successfully connected to FWD with IAX2 and to GIZMO using SIP. Just purchased a Motorola Wildcard X100P and installed it into a clone PC running Fedora Core 6. Analog

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware

Anybody has figured out why asterisk + snom have such bad quality using GSM? When I use GSM I see such messages dumped on asterisk console: WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect

Hi Asterisk-ians! Need all of your help. I am stuck with this issue for last few days. I have one X100P installed in my system. My Asterisk is registered with another Asterisk Server/SIP provider as client and the call is successfully received by my Asterisk server (named as starwars). Now, the extentions.conf file states, the incoming INTO * should go out to fxo as below: exten =>

Hi, Anyone knows a software fax device which can act as a SIP UA? I want to have a SIP based FAX machine (sofware) on a PC associated with an Asterisk extension. Thanks, Dan

I am still having unacceptable echo on my X101P and twidling with the rx/tx gain levels and echo settings appears to have no discernable effect. Some questions for those who may have more significant electrical engineering background than I. 1. This impedance match thing ... will it affect this solution having other phones in parallel with the X101P? This is done so that I can test while not

I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i

Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal

I'm sure dumb when it comes to describing things that happen on my system. I'm making an outbound call on my ATA186 when another call comes in. I first get the nasty CID screech and then the periodic call-waiting tone. So far, so good. Then I hookflash, and just like it's supposed to, the waiting caller is on the line. But during the duration of that conversation, my console

So I'm not getting caller ID via the X100P card. I've confirmed the PSTN is sending me caller id by plugging in a little third party box. Ideas, tips, greatly appreciated

Hi Guys, Pardon me for the might be stupid/repeated question ( I am new to the game), but has anyone successfully implemented any FXO cards in Australia. Particulalry I am thinking about X100P card. Is it good for local lines? Thanks for your advice.

Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,