similar to: What would be required for this?

Hi we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to

I am using 2 X100P clone cards in my asterisk system, and have one phone line plugged into the 'line' port of each card. On one of the cards, I also have a cable plugged into the 'phone' port and this cable is then connected to the modem on my PC so that I can use it for dial-up. I thought that this would then allow me to do dial-up and at the same time it would lock the port so

Hey all, Wondering if this is possible.. Incoming call is answered through X100P, then an extension is dialed using the same X100P card. Basically I want to dial in, enter 9 + <phone#> and have it do a flash then have it dial *08 <the same phone number> + # on the same PSTN line to have it transfer my call to another phone number. I realize this isn't very safe, but I would

<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P> <P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P> <P>I've tried copying the config in this listing with no success. </P>

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has

Hey all, ? I'm trying to get an E400P and 2 X100Ps working together in the one box and don't seem to be having much luck. ? I can get the two different types of boards working separately, but not together.? I've made calls on both the X100P and have seen sync and correct signalling on the E400P.? But when I try to enable to configs together I get the following: ? modprobe wcfxo

If anyone is looking, I just ran accross an ebay auction for X100P Cards at what I thought was a very reasonable price. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3046843672&category=48483&rd=1 --------------------------------- Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software -------------- next part -------------- An HTML attachment was

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

I'm trying to spec out hardware for a new office, and I'd like to include power over Ethernet as an option. I've seen a handful of PoE injectors around $1000 for 24 ports and a couple switches up around $2500 for 24 ports. Are there any cheaper options, short of buying a boatload of 1-port injectors off of ebay? I don't really need more then 24 ports of PoE out of 48 total

Thanks for the reply Bill. I am aware of the interrupts problem. To solve it I have already disabled my serial ports freeing up interrupts 3 and 4 and these are allocated to the two cards. This was done 2 months ago and has not solved the problem. Is there any way that something can wake up every now and then and generate these two interrupts? My current /proc/interrupts is as follows:

Hello folks, I'm very interested in the Digium/Asterisk combination but need some clarification. I would like to setup a SOHO for business and home use. Scenario One: I have one analog line, 4 analog telephones. Do I need a TDM400P + 4 FXS modules (Green) + X100P? Scenario Two: 2 analog lines, 1 selective ring number for fax, 8 analog phones. Is this what I need? 2 TDM400Ps and 8 FXS

I am interested in running asterisks and gno phone. However, i have yet to purchase the x100p cards. Is there a way for me to test asterisks in like a simulated mode or something? (to get used to it and the configs and stuff, prior to purchasing the x100ps?) Thanks all, A.J.

Hi Most of the threads in the list archive relating to X100P and hangups are about not detecting hangups. We have got the opposite problem. We have experienced an increased number of false hangups when connecting an X100P to an analog port of an ISDN terminal adapter. It happens more frequently on incoming calls than it does on outgoing calls. Often hangups occur after about 3-4 minutes into the

List members, Is it possible to record outbound analog calls using an X100P? I was asked if I knew how to record all calls for a shop with 4 analog phones transparently to the end users. I thought Asterisk was a good fit for this and I envisioned using either Digium TDM400Ps or Sangoma A200s with 4 FXO and 4 FXS modules. The FXO modules would be connected to the existing PBX and the FXS

Yes and No The D500 is a terrible thing First problem: it sends some horrible DTMF, so if your voicemail is configured to send #H and *H, it will not work, configure it to send numbers, like 8H and 9H (H is a placeholder for the extension). I also managed to use the MWI (message light), it's a perl script that is in voip-info.org, but with a little correction because the wiki distorted it. If

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

It would appear that the "uniqueid" field is not being populated in the MySQL CDR DB.. Is this an obsolete field or is a bug? Later..

Hey all, I'm trying to get my Asterisk server up and running on fwd.pulver.com just to get the hang of it until I get my FXO card in a couple of days. It seems to connect but that's about it. If I try to dial into it from another fwd # it says user is not online. In sip.conf I have the following added: register => xxxxxx:xxxxxx@fwd.pulver.com/489125 [fwd.pulver.com] type=friend

Hi I have created a Mac OS X installer package for installing Asterisk on OSX ver 10.2 and 10.3 Anyone who'd like to give this a try, please download the installer package from here ... http://www.astmasters.net/stuff/Asterisk.pkg.tgz to install Asterisk on OSX just double click the package file. please send any feedback to benjamin (at) sunrise (dash) tel (dot) com NOTE: this is a

The problem I mentioned a day ago is now getting worse. message log: kernel: PCI Master abort kernel: Unable to handle kernel paging request at virtual address 00230004 kernel : printing eip c02060a6 kernel : Oops: 0002 kernel : CPU: 1 kernel : EIP: 0010:[<c02060a6>] Not tainted ,... Some register dumps kernel: Process asterisk (pid: 23917, stackpage=c5aff0000) kernel: Stack: numbers:)