similar to: Maximum number of X100P cards in the same * box

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Gary Franczyk > Sent: Thursday, January 15, 2004 10:37 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question > > > Whaaa?? So, to allow 24+ lines of dial in access, how

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

G'Day, I need from regarding using the X100P. Some are specific to Australia and I have been told in IRC there are already existing users in oz, so I would appreciate your input. 1) Is the X100P Australia telco approved? 2) I have setup asterisk with X100P and whenever I dial in I cannot get any callerid information. Is this standard behaviour? If not how have other australian users managed

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

This might be a newbie question but I'm just wondering how would it be possible to have 30 analog lines using asterisk for PBX by just using TDM40B and X100P (or are there any device>), if an ordinary PC support just 4 PCI slots? the maximum scale i guess would just be 2 x 8. Adding a new PC just for this purpose would be costly. I would appreciate your comments. Thanks.

Assuming the price of an ADSI screen phone (say, Aastra 390) was the same as an IP screen phone (say, Cisco 7960) and someone was setting up an * server for their 20 employees (each of whom would have either an ADSI or IP phone on their desk), would there be advantages to using the ADSI phones over the IP phones, or vice-versa? For discussion, let's assume that the hardware needed to

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of David Burr > Sent: Sunday, January 11, 2004 4:31 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive > > > We have a new contest starting today! > > The first three

> -----Original Message----- > From: Travis Johnson [mailto:tlj@ida.net] > Sent: Monday, September 08, 2003 1:05 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7940/7960 ethernet ports > [...] > We are having a problem with Cisco 7940 and 7960 phones when > the PC is plugged into the 2nd ethernet port on the phone. It > will drop the

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

....I THINK. When dialing 1+10 digits, I occasionally get a telco message "You must first dial a 1....". When I look at the console, the number is being sent to the ZAP channel properly. We're talking about a couple of POTS lines on a TDM400P. I'm thinking that it may be starting the dial too early after coming off-hook because I can just redial and have it work (or not)

Just starting to do the research on this one....I've got a customer who is showing interest in replacing any older Panasonic unit providing service to 96 tip/ring lines from a single PRI. Does anyone have any recent experience with a decent (as in, plays nice with * and has a reasonable per-port cost) channel bank or similar? Mediatrix only goes up to 24 port, as far as I can tell, which

I'm a bit displeased at the way this happened. I received an email from VoicePulse. Here's some excerpts: ------------------ >We're sending you this important update so you can take advantage of improvements we've >been making to your VoicePulse Connect! service. >We've been working hard on improving the audio quality and reliability of your Connect! >service,

Im trying to figure out the base way to check the callerID being sent to my Asterisk box and use it if it is a valid NANP number, but replace it with a static NANP number if it is not. (Why? I have a few carriers that require this, and a few international users - if it happens to take one of the carriers that require it, I want it to set a static number that is valid). I'm playing

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Jeremy McNamara > Sent: Thursday, December 11, 2003 2:19 AM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Re: * with RADIUS > [...] > > Explain why you think you really need RADIUS Accounting? >

Hello, I have ISA card LineJack. I could not find any information if this card can work as fxo with Asterisk. If it can work, can somebody point me how to install it on my Asterisk box. Or maybe there is some documentation about it how to install LineJack. I will be very thankful for any help. Regards Bartosz Jozwiak ------------------------------------------------- This mail sent through IMP:

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

Here's the latest batch of words to get shipped out to Allison Smith. Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern time, and I'll add them. As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying for this round out of my pocket. Please send to paypal address "jtodd@loligo.com". I did not include all

I am working on a call-back solution where the initiating call should never be answered. I was doing this simply through the dial plan, sending a progress tone, and then dumping the channel, and firing off a DeadAGI which created a call file to make the callback. Now I've tried extending this so that an AGI is fired first to check for things - like no inbound ANI - and play a

Hi, I had ordered a TDM40B and developers kit a few months ago. I have everything installed and working, with one exception - sound quality. When placing a call it sounds like a very bad cordless phone - lots of hiss / static in the background. This even happens with the dialtone, though it is much worse one the call is connected. This does not occur when the phone is directly connected to the

I'm looking for an application that can monitor a channel for voice input and then proceed on. The closest thing I've found is BackgroundDetect, which expects DTMF. Here's what I'm doing: -Call file generated which calls someone and connects them to an extension. -Extension plays stuff, etc. etc. etc (not important) With digital or VoIP termination, this works fine, because *