similar to: indications.conf settings for spain

Hi everybody, I had an ISDN card with winbond chipset and isdn4linux, but there was a lot of echo when call from SIP to ISDN, so I buy an HFC chipset card (Conceptronic c128i). I downloaded and compiled bristuff-0.2.0-RC5 and everything is going fine. The sound quality is excellent, there is no echo, but i have a strange problem. Sometimes calls made from SIP phones to ISDN just finished

Hello to all: When I make a call of extension (FXS) to extension (FXS) (TDM40B) status of the call is the following one: #### Case call answered --------------------------------------------------- Log init -- Starting simple switch on 'Zap/4-1' -- Executing Macro("Zap/4-1", "stdexten|103|Zap/2") in new stack -- Executing Dial("Zap/4-1",

Hello list: If I am not mistaken the module 'ixj' is for the cards 'Quicknet LineJack' is possible not to load it when starting asterisk ?. How call asterisk this module ?. -- Jose M? Guisasola Consultor T?cnico CMSI 2002 S.L.

Hi Vic, I hit that same problem! My SIP phones would sound okay when I made changes to indications.conf but incoming calls in to my TE410P had their own thing going on! Have a look at the zaptel source files, there's one called zonedata.c. You'll see the au settings... replace what's there with this: { 1, "au", "Australia", { 400, 200, 400, 2000 }, { {

In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au

Quisiera saber si alguien me puede ayudar con documentación o el lugar donde encontrarla, relacionada con la elaboración de paquetes en R, tengo algunos script programados y quiero convertirlos en paquetes; he leído el manual para hacer extensiones pero quiero algo más sintético y practico,  Agradezco cualquier colaboración al respecto Heber [[alternative HTML version deleted]]

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

Hello asterisk-users, Just curious if anyone has the indications for Ireland, tried googling for it to no avail. Sean -- +---------------------------------------------------+ |VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie | |GPG Key http://thecivvie.fastmail.fm/thecivvie.asc | +---------------------------------------------------+ Strange things happen under the midnight sun when Men

I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880",

Hi All, I am trying to reproduce this using R instead. [image: Full-size image (38 K)] I tried using the following code *SChla <- read.csv("SM_Chla_data.csv")* *Atlantis <- SChla[16:66,]* *head(Atlantis)* * * Seamount Station Depth Pico Nano Micro Total_Ch dbar Latitude Longitud 16 Atlantis 1217 Surface 0.0639 0.1560 0.0398 0.2597 2.082 -32.71450 57.29733

My telco doesn't provide Disconnect Supervision or Polarity Change. So I figured I have to detect hangups with busydetect=yes in zapata.conf. I tested it. When the telco sends a busy tone * detects it and hangsup. So far so good. The problem is the telco doesn't always send a busy after remote hangup. Most of the time it sends a congestion tone. I am guessing these tones from what I

Hi i have the data in the following format: rent,100,1,common,674 pipe,200,0,usual,864 car,300,1,uncommon,392:jump,700,0,common,664 car,200,1,uncommon,864:snap,900,1,usual,746 stint,600,1,uncommon,257 pull,800,0,usual,594 where as i want the above 6 lines data into 8 lines as below (Spliting row 3 & 4 at : and sending to a new row): rent,100,1,common,674 pipe,200,0,usual,864

Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937]

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

I have an hp9000/c160 with hpux10.20+gcc2.7.2. When I try the example in the ace help I get: > library(acepack) > ?ace > TWOPI <- 8*atan(1) > x <- runif(200,0,TWOPI) y <- exp(sin(x)+rnorm(200)/2) a <- avas(x,y) /usr/lib/dld.sl: Unresolved symbol: pow_dd (code) from /home/buyuk/R-0.50-a1/lib/acepack.so ABORT instruction (core dumped)

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 186 bytes Desc: Signierter Teil der Nachricht Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040624/fe4cdcbf/PGP.pgp

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 =>

Hey All, Having a bit of a problem with a Wildcard X100P card. When I try to make an outbound call using the card, it picks the line up and then only dials a single digit. I've confirmed it's only dialling a single digit by listening on a phone plugged into a parallel socket. Incoming calls work fine, it's only outbound calls I'm having a problem with. I've tried

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 =>

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 =>