similar to: Softphone vmail indicator and TDM400P woes

Since the irc channel wasn't any help, I will try posting my problem here. I have two TDM400Ps less than a week old in a PC. All of the FXS ports work fine, and all of the FXO ports worked fine up until thisafternoon. If I try to dial in, I get a busy signal, if I try to dial out, all I hear is a very scratchy, very crackly dialtone. If I swap the first FXO module with the second on the card,

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

Hi all, Theoretically, I know it's possible, but is any using multiple tdm400ps (fxo) in single * box? In a production environment? Any gotchas aside form irq sharing? Thanks Ryan

Hi all, haven't seen many posts about asterisk in Singapore... Getting a server going there and was wondering if TDM400Ps will be fine in FCC mode, and if there are indications / cadence values that I should be putting on there as in other international locations. Seen an unsettling post on voip-info about Singapore issues with Call Polarity/Hangup Detection -- crossing my fingers I

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

Someone mentioned earlier (I can't find the message now) that they had a motherboard that allowed you to change IRQ assignments in BIOS. Does anyone happen to know how to identify motherboards that can do this? I'm going to put together a new machine now and I'm having trouble picking a motherboard for it (ordering from Dell or other online vendor is not an option, since I need

Hello Asterisk Community! There have been some complaints made by those customers that purchased the TDM400P board and it didn't work properly in their boxes. Digium promised to swap such boards for the new - revised version and will keep the promise. However since we were backordered we're currently shipping boards to the customers that paid and are waiting. But if you want to receive

Hello. I've seen several posts talking about line quality using Digium cards that are sharing IRQs or on machines where X is running but after trying all of those fixes I am still having a problem with line static on outoing calls. BTW, calls that are from one extension to another extension have no static, however, they have occasional clicks and pops. At any rate, I was wondering if

If i leave my asterisk server running for a long time then try to dial outbound on the zaptel channel i get this high pitch static noise and won't dial out. This behavior is happening over two different servers i am using. Rebooting asterisk does not sovle the problem. I rmmod the zaptel driver then reload and that solved the problem. But i cannot continue to do that. Also sip to sip

I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is any registerable incoming volume from these lines. I've been running them at rxgain = 25 (zapata.conf) to make the audio audible, however this creates poor call quality issues

G'day list, It appears I got ahead of myself in believing I'd solved my CND (caller id) problem. My callerid.c change got it working on one of the handsets I have, but not the other. (Recap: The AU CND standard has an increased delay between the end of the ring signal and the start of the CND FSK data -- it's 500ms in US GR-30-CORE, and around 700ms in AU DC.002/TS030). I've

I''ve pared my rjs experiment down to what seems a bare minimum. When I click on my link_to_remote, the rjs executes, returns the correct javascript to the browser, but it doesn''t update the div I''ve specified. What appears to be coming back from the controller is: Element.update("my_form", "hello, rjs"); I''m tearing my hair out.

I have a simple barchart with horizontal bars and horizontal tick labels, produced with barplot(x, horiz = T, names.arg = c, las = 1) The labels are longish strings, truncated on the plot. I wish to leave more space for the left margin, and experiment with mar parameter, barplot(x, horiz = T, names.arg = c, las = 1, mar = c(5, 15, 4, 2)) trying various values for the second vector element, but

Removal of ans[nas] <- NA from the code of function 'ifelse' in R is not committed (yet). Why? -------------------------------------------- On Mon, 28/11/16, Martin Maechler <maechler at stat.math.ethz.ch> wrote: Subject: Re: [Rd] ifelse() woes ... can we agree on a ifelse2() ? Cc: R-devel at r-project.org, maechler at stat.math.ethz.ch Date: Monday, 28 November, 2016, 10:00

Hi, I have just built my home Asterisk box into a better PC that became available (still only a P2 350 but it only has to manage 1 analog line).. Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P (1 module installed).. These cards were working fine in my older PC that was running my Asterisk at home.. The inbound calls via the X100P to my sip phones are working great..

Anybody notice log delays in this list, and very small amount of traffic? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/576a9b0e/attachment.htm>

----- Original Message ----- > From: "sean darcy" <seandarcy2 at gmail.com> > To: "Asterisk Users Mailing List, Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Tuesday, January 21, 2020 9:22:28 PM > Subject: [asterisk-users] permission woes on systemd [..] > So why would starting asterisk as user asterisk work, but fail using

> Message: 7 > Date: Mon, 15 Nov 2010 16:45:34 +0000 > From: Timo Sirainen <tss at iki.fi> > Subject: Re: [Dovecot] proctitle woes with 2.0.7 vs. 2.0.6 > To: Clemens Schrimpe <csch at kiez.net> > Cc: Dovecot Mailing List <dovecot at dovecot.org> > Message-ID: <1289839534.1764.138.camel at kurkku.sapo.corppt.com> > Content-Type: text/plain;

Why m3u vs. pls? Brian ----- Original Message ----- From: "harvey smith" <harvey@buskers.org> To: <icecast@xiph.org> Sent: Monday, September 10, 2001 12:49 PM Subject: Re: [icecast] Client Woes > On Mon, 10 Sep 2001, Jack Moffitt wrote: > > > The way to prevetn this download, or rather to work around it, is to use > > a small intermediate playlist file.

Related to the length of 'ifelse' result, I want to say that "example of different return modes" in ?ifelse led me to perceive a wrong thing in the past. ## example of different return modes: yes <- 1:3 no <- pi^(0:3) typeof(ifelse(NA, yes, no)) # logical typeof(ifelse(TRUE, yes, no)) # integer typeof(ifelse(FALSE, yes, no)) # double As