similar to: TDM400P slow getting line tone

I'm having slight problems with outgoing audio quality on Zap channels. People hear an interrupted voice. Can anyone help..? Regards, Fabrizio Mazzoni Macron SPA

Hello all, I have a problem that i cannot resove so I'm seeking for some help. Basically i want a user to dial from the PSTN line into our * box, dial a certain extension, give him a voice propmt asking him to insert a number and store this number. I need to do this because i want to give the possibility to our employees to dial to the free voipbuster countries from home so that they do not

Here is my system: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard) CPU AMD|2500/333 ATHLON XP BARTON R(Standard) DDRAM 256M|DDR333 PC-2700 -K %(Standard) HD 40GB|WD 7200RPM 8MB WD400JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956 RTL(22) Besides the above parts, i have 2 X100P FXO PCI cards, and 1 TDM40B I bought 2 phones from radio

Hello, everyone After having spent several time to look for any solution for my problem, I decided to write here. Here is the problem got a Digium X101 FXO card and Digium TDM400P alias freshmaker (1 fxs module on it on first port ) in my asterisk box. The X101 works perfectly. The problem comes from the TDM400, when i plug a tested working phone on it , i get no dialtone; the goal is to

Hi All, I have the problem setting up TDM22B card. Steps what I have followed are: [1] compiled zaptel-1.0.9.2 & installed the same. [2] modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg

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..

Hello, Does anyone know what may be causing this? Asterisk was built from cvs tonight. Ztcfg also says us is an invalid tone zone. Anyone got some information on what this is, why is it happening, and possibly some solutions? [root@anistonetech zapata]# asterisk -f -d DEBUG[1024]: File config.c, Line 653 (__ast_load): Parsing /etc/asterisk/asterisk.conf DEBUG[1024]: File config.c, Line 653

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,

Hopefully someone here can save my sanity. I have been trying to solve this problem for days now, but just cant put my finger on it. Im new to * so I have probably done something stupid! I have a TDM400P with one FXO module and a FXS module. The main problem I have is not being able to get the extension attached to the FXS module to ring or be able to make calls. It gets a dialtone fine but I

I installed the TDM400P and installed it on a system running on Fedora Core1, please refer to the steps below. I connected an analogue phone to the FXS port. When I pick up the speaker I don't hear the dialtone although when I press a number key I hear a DTMF generated. What could be the problem? I appreciate your help 1- First I installed the TDM400P in the PCI slot and connected

Dear R-list, I have two variables (numerical and categorical) and would like to have the sum (and maybe some other statistics) of the numerical variable by the categorical one. Can you help me, Thank you, Fabrizio --------------------------------------------------------------- Fabrizio De Amicis IT Department Generali Information Technologies - (GIT) Centro Galleria 2, Via Cantonale CH -

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

HI Asterisk Gurus I'm connecting my * server to a zplex 10B, I'm using a slot on my TE410P card as T1 to connect my server to the channel bank (i use a cross cable), the zplex doesn't show any alarm neither the server. I'm trying to make calls from (ext 1) from the channel bank, to other (ext 2) in the same channel bank, but when I dial the ext number, the dialtone doesn't

Hi, I'm having a problem calling international numbers with debian's Asterisk 1.0.7 w/ zaptel 1.0.9 in Moscow. Russia doesn't seem to have touchtone dialing, so pulsedial is enabled on my TDM400P interface. Local numbers work fine, but when it comes to long distance or international, I'm lost. The prefix for these should be 8 (wait for dialtone) 10 (country code) (city code)

Hello all, I have a problem with overlap dialing and don't know how to get rid of it. My setup is: 1 HFC card with bristuff -> ZAP/g1 (2B + 1D channels), SIP phones (I just removed TDM400P with 4 FXS) I created test extension 222 which goes directly to g1. In Zapata.conf overlapdial is set to yes. First I created this extension: exten => 222,1,Dial(zap/g1,100,tc)

How I change my configuration to reduce this issue. I have this setting on my zapata.conf signalling=fxs_ks group=1 callgroup=1 pickupgroup=1 channel=1 signalling=fxs_ks group=2 callgroup=1 pickupgroup=1 channel=2; singalling=fxs_ks group=3 callgroup=1 pickupgroup=1 channel=3; singalling=fxs_ks group=4 callgroup=1 pickupgroup=1 channel=4 and for outbound calls I have this context on my

Another easy(/stupid) question: with the following command j2<-xf1[1:10,"V4"] I have > j2 [1] CHROMOLI LINEAGERMAI RINALDI GIUNTIMA AUTOSTELLA [6] CAIZZONE CENTRO B PEL E C SNC CONSORZI MAN NORD PDM 1304 Levels: MACHIAVELLI Snc MENARINI MANUFACTURERS LOGISTICS SERVICES ... ZUCCHERIFICIO

I think I've configured everything to have stuttered dialtone on my analog phones (it works fine for my SIP phones). But I still don't have it:-( I'm using asterisk 0.9.0 and zaptel 0.9.1. In voicemail.conf: [local] 21111 => 987654,Robert Withrow,email@here.com In zapata.conf [channels] ... mailbox=21111@local channel => 1

Hello to all, I've a frustrating issue, everytime I'm trying to connect to some website (like Linkedin, VMware, CertCollection, etc etc) I have issue with credentials... Or the website didn't recognize my login (and I'm still in login page) or for example I'm logged in linkedin but I'm not able to see which visit my profile cause it ask me my credentials and when I enter

Dear R.List, I am starting to use R. I have an easy question. In a dataset of 15 variables, I am not able run correctly the index i of the do loop. Do you have any suggestion? NC <- function(x) for (i in 1:15) { print(dim(table(dt[,"Vi"]))) } Thank you in advance, Fabrizio --------------------------------------------------------------- Fabrizio De Amicis IT