similar to: X100P Clone

I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC

Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the

Hi all, I just started programming using AGI and I have a simple question about STDERR. If I understood it right, all the messages sent to "STDERR" should be shown in the Asterisk console, but using the following python code I just can't see anything. #!/usr/bin/python # # File: /var/lig/asterisk/agi-bin/agi-test.py # # Description: An AGI Script # import sys env =

Hi all, I just want to know how I can make sure that incoming calls to my asterisk server are being treated by [incoming] section of extension.conf file. Thanks in advance. Ronaldo.

Hi all, I have a IAX trunk between two asterisk servers, both with dynamic IP and both have a DNS name associated with it. In the iax.conf file I configure the "host" parameter with the DNS name of the servers. Everything works fine until one of these servers get a new IP, so the other can't find its peer (the one that has just gotten a new IP). If I manually issue a "iax2

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...

Hi , > > My Xen has been working fine for a few days. Then today it suddenly > can not run. > > When it runs It give me this error : - Error starting domain: virDomainCreate() failed POST operation failed: (xend.err "Error creating domain: Boot loader didn''t return any data!") Details: - Traceback (most recent call last): File

I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf files but I can't see anything that I've missed. Cheers Matt

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...:-))

Now that the X100P is no longer being offered by Digium, what is the best solution? I seem to have run into a few posts where people talk about problems they've had with their X100P clone cards (dropping calls, echos, etc) other people seem to not have any problems. Of the three chipsets that will work: Intel 537EP, Ambient MD3200, and Motorola 62802 (as seen here

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

Does anyone know if the X100P clone cards provide the timer needed to run MOH and the Conferencing service? I have no need for a T1 card, but I'm running asterisk on a dual processor machine with the wrong kind of USB devices, so none of the dummy timers will work for me. Also, I'm running Asterisk@Home (2.4 Kernel). Anyone know if upgrade to the 2.6 kernel will allow me to use the

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 =>

I'm new to asterisk and fror a cupple of days I heave been googleing the net for digium "clones", because it's very hard for me to get a digium card (X100P). Does anyone Know another substitute for X100P (I know that intel based modem with chip 537/MD3200 is working but I did not find any of those) ? I made it work with an Intel 536 (with a costumised driver that I found on

Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up

I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [root@asterix root]# modprobe zaptel [root@asterix root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel

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

I'm not sure if I have a configuration problem or not. I am unable to dial out. When I try to dial in I can hear the phone ring on the dialling phone but Asterisk does not register anything. In zaptel.conf I have loadzone = au defaultzone=au fxsks=1 In zapata.conf language=au context=from-pstn When I do: zap show channels I get: Chan Extension Context Language

Has anyone had any success with this card? Thank you. Sandy.

Hi, I'm cutting my teeth setting up a home asterisk server, but having some trouble getting Zaptel to recognize a X100P clone PCI (Motorola SM56 which should work according to various wiki's). What struck me was that the "/proc/interrupts" do not include IRQ 11 as the "lpcpi -v" is reporting. Am I missing something fundamental here? Any hint is greatly