similar to: fxo/fxs cards with 8 ports

hi, To post, and to reply to a post, i have to goto my email. Now, if there is a web interface to these mailing list, things would be easier.

Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is no a forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong. That is why I will post my questions here: 1- Transcoding: is this when

I have a file with 3 cols as: x y value 1 1 0.5000 1 2 1.8333 1 3 4.3333 2 1 0.7500 2 2 0.7500 2 3 4 3 1 0.7500 3 2 1.4167 3 3 5.1667 I read them in using p <- read.table("data.txt", header=TRUE) My questoin: how to draw a image to show the values at different positions (x, y)? I would like to see different colors for

I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

Hi, Small questoin: when running the P2v hard disk image, and I want to convert a phyical machine to oVirt, to which machine do I connect when the GUI starts? one of the nodes or the hosted engine? (i'm not talking about the URL). Thanks

diff -u -r -N openssh-2.9p1/channels.c openssh-2.9p1-ajf.1/channels.c --- openssh-2.9p1/channels.c Tue Apr 17 14:14:35 2001 +++ openssh-2.9p1-ajf.1/channels.c Wed Jun 6 23:25:36 2001 @@ -1815,13 +1815,25 @@ /* * getaddrinfo returns a loopback address if the hostname is * set to NULL and hints.ai_flags is not AI_PASSIVE + * + * Oh yeah? Setting hostname to NULL and hints.ai_flags to +

Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN <--> Asterisk <--> E1 cable <--> Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I

Hi, I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a TDM2400P with echo canceller. I installed the card but no echo cancellation is being made...seems like the echo canceller module does not work, infact the software cancellation is working. My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but no echotraining parameter which gives a warning. I found

Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5 im having an error message when in running asterisk with the tdm card in. here's the error from the console of asterisk: [Nov 18 10:30:44] ERROR[5557]: chan_zap.c:7489 mkintf: Unable to get span status: Inappropriate ioctl for device [Nov 18 10:30:44] ERROR[5557]: chan_zap.c:10466 build_channels: Unable to register channel

Hi, I'd like to change 3 TDM400p with one TDM2400p to avoid echo (my intention is to use a TDM2400P echo cancel module). It TDM2400p working good with asterisk 1.2.1? Or I need to install a new asterisk version? TIA Giorgio Incantalupo

We just bougth a tdm2400p with all the modules for FXO, but we are having some troubles with the card, cause it aparently is stripping some digits from the dialed number, we tested the same server with a tdm400 and everything worked as expected. We?ve already added "w" before the dialed number with no results, is there any way to solve, is it a bug thanks

Hmm, I guess I need a sys::Math::getRandom() function that uses a "good" random number generator on the given platform. I'll make a note of this and tuck it away for future implementation. Reid. Chris Lattner wrote: > On Tue, 28 Sep 2004, Paolo Invernizzi wrote: > > >>The next major problem is that VC has only >> >>int rand(void) >>void srand(

I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot

Here I am again... I missed that in the previous diff, they are the usual missing <algorithm> and the std:: namespace missing in sort, find, make_pair. Alkis can you please give them an eye when you have time? The next major problem is that VC has only int rand(void) void srand( int seed ) So I donno how to compile the ExecutionEngine/Interpreter/ExternalFunctions.cpp that refers to the

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Thx, Juan __________________________________________________________ Mensagem enviada usando o Webmail da ViaLink ver. 2.7.8

Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. The digium card running on Intel 915G chipset. Below are my zaptel

Hello. I have a TE212 configured in E1 mode. This is shown in a cat /proc/zaptel/2 and 3 (where the card is configured): cat /proc/zaptel/2 Span 2: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" HDB3/CCS/CRC4 RED NOTOPEN 25 TE2/0/1/1 Clear 26 TE2/0/1/2 Clear 27 TE2/0/1/3 Clear 28 TE2/0/1/4 Clear 29 TE2/0/1/5 Clear 30 TE2/0/1/6 Clear

Hi, I am getting half channel audio on cisco 7960? Any idea why? Details below. Jerry This same phone has been working for MONTHS using a TDM2400p with no issues. Today we got a T1 installed coming into Box A with all incoming calls going to Box B (the TDM2400P box). The TDM2400 is no longer in use. All incoming and outgoing calls are done with the T1 and Box A and Box B use SIP between them

I've been asked to assess the cost of implementing Asterisk with a single T1 line in one of our offices. I've had plenty of experience w/ TDM400 cards, but T1 is new for me so a couple of questions: 1) Will I need a digital or analogue interface card? I expect digital is the answer, but the Digium web site said something about analogue cards being able to support "provider T1