similar to: Timing, use analog card, ZT Dummy etc.

Anyone have any idea if there is some sort of limitation to the number of SIP or IAX end points which can register to an Asterisk system (2.8Ghz dual processor, 2GB ram) while also handling 30-50 simultaneous calls without getting into trouble? Of course the 30-50 simultaneous calls end up being 60-100 channels of mostly G711 VoIP. We have seen issues where our Asterisk just gets all crazy and

Hello, I ordered the Devel lite kit, and installed it. I am just trying to get the FXO port to work, and am having trouble. To load the card I do the following. modprobe wcfxs modprobe wcfxo ztcfg -vv asterisk -vc My /var/log/asterisk/messages show Nov 3 11:03:39 WARNING[3317]: zt hook failed: Device or resource busy Here is my /etc/zaptel.conf fxoks=1 fxsks=4 loadzone=us defaultzone=us

Hi list! I'm totally fed up with bristuff (or it's instability with a simple HFC-S card), 2 out of 3 times when people try to call they get the information tone that the number is not connected. I would like to try vzaphfc and I am looking for information on it. >From previous posts I found that the only place where the sources seem to be maintained and available is at the debian

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten =>

We are looking to be able to put a device in front of an array of Asterisk systems which would do the job of load balancing them. We would store all the particulars on one or more MySQL servers. What want to accomplish is to have all calls sent to/from a single IP, then push the calls off to another Asterisk server in the array. If one server goes out, we are hoping there will be no effect other

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I have always been able to block toll-free numbers by catching them with a line similar to this for each DID I have on my system: exten => 5554441212/_888NXXXXXX,n,Playback(GoAway) Where 15554441212 is one of the DIDs that rings into our Asterisk box. The problem with this approach that I have to create a line like this for every pattern I want to block multiplied by every DID on my system,

Hi, I've been trying to bring our Asterisk server to the latest version. I've grabbed the latest CVS and upon trying to compile zaptel, I get the following errors: gcc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c gcc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB

Hi all, I have a TDM400 with all FXO on it. When I make an outgoing call, I can hear callee but callee claims the volume is too low so that he/she can't hear very clear. Can I adjust to increase the volume in callee side? Is it increase the value of txgain can solve the problem? ango

Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX-XXXX 3- Anything else that should be restricted if one was to restrict all calls to US 48 only I have found many list but it's tough looking at the entire list of area codes and pulling out each of them

Hi, I'm wondering why zttest shows Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469 Shouldn't it be 100% as timing is hardware and comes from PRI? Am I missing some kernel config? Regards, Atis My /etc/zaptel.conf is span=1,4,0,esf,b8zs span=2,3,0,esf,b8zs span=3,2,0,esf,b8zs span=4,1,0,esf,b8zs #lspci 07:03.0 Communication controller: Digium, Inc. Wildcard

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

Hello All, Does anyone know of a software emulator that can be used to simulate hardware such as an E1? I need to play with AstUnicall in a test environment and don't have access to these circuits from the US. If there is an alternate way to test/play with AstUnicall, please let me know! Thanks, Mark -------------- next part -------------- An HTML attachment was scrubbed...

Asterisk 1.2.7.1, Red Hat 9.0, TDM 400 2 FXS, 1 FXO I'm getting the following warning from time to time: "zt hook failed: Device or resource busy" It seems that once I get this error I can no longer use my zap channel. Interestingly it seems to affect SIP as well as I can no longer dial out on that either, it begins to complain that my GSM licenses are used up (I only have

Hello, I am using 1.4.8 and have a question about Queues. I noticed that if I have an agent logged in using AgentCallBackLogin and that agent is unreachable for some reason (SIP phone unplugged) calls to him/her will completely yack. For example: 1-Agent 500 is the only one logged into queue number 1. 2-A call comes into queue number 1 3-The call is pushed to agent 500 at extension 21 which is

Hello, I'm trying to develop a box plot of time series data to look at the range in the data values over the entire period of record. My data initially starts out as a list of hourly data, and then I've been using this code to make this data into the final ts array. # Read in the station list stn.list <- read.csv("/home/kbennett/fews/stnlist3", as.is=T, header=F) # Read in

Does anyone have any suggestions for connecting analog DID trunks? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4

I know it is a strange arrangement but due to contracts, it is what it is, no PRI for now. I wonder if anyone on the list has run a server with both types of cards installed? Results? I have never touched a BRI except in concept and Cisco lab. Not sure what the BRI stuffed package may or may not do to anything else that might relate to zaptel or Sangoma. Thanks, Steve Totaro

I'm running dahdi on the host system, and have added the /dev/dahdi/ devices to the guest vserver as recommended in Beave's "Virtual Private Asterisk" whitepaper (http://www.telephreak.org/papers/vpa/). I tried copying libtonezone.so and libtonezone.h to the guest, but I couldn't anything to replace zaptel.h in DAHDI souce (it seems dahdi.h was deprecated?). I need to

Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound. What can cause that problem? Thanks in