similar to: Inbound Voice Quality - Speed Change

I'm struggling to get my VOIP installation to be acceptable. I'm looking for advice on what else I can look for. My system: o Teliax VOIP service, voip-ny1 proxy o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms average jitter) o 3.2 GHZ P4 Server (runs asterisk, firewall, other stuff) o server lightly loaded o Linux kernel 2.6.19.2 o Shorewall Firewall software with

I''m having troubles with my outbound VOIP connection. I''m convinced that I don''t have QOS/traffic shaping configured properly in my shorewall linux firewall, which serves as my Asterisk VOIP server and Internet router/gateway. I don''t have a separate router box. I''ve been using VOIP for about a year now, but just recently realized that I need to

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

I've upgraded to asterisk-1.6.0-beta2. I'm trying to get the new Telco MWI detection function working. It doesn't appear to be working. I have this in zapata.conf ; PSTN connected here ;immediate=no ;busydetect=yes ;busycount=8 ;musiconhold=default mwimonitor=yes ;mwilevel=512 mwimonitornotify=/usr/local/sbin/zapnotify.sh faxdetect=incoming signalling=fxs_ks context=incoming

How do I get a context to continue to execute commands after the caller hangs up after a Dial command? I'm using the "e" option to the Dial application. I though the "e" option would allow the context to continue. This doesn't want to work for me. I'm using asterisk-1.6.beta5 I never get to "3" below. I get a message saying the "2" ended

I use Asterisk in my house. Each phone is a different extension. I really like the ability to have multiple simultaneous calls in the house. However, I do miss being able to be able to pick up a phone in a different room. Currently, I have to either transfer the call or transfer the call to a "conference" extension to move around the house. While a connection in progress on one

> he graphics card is emulated in Qemu, the graphics operations are > converted to VNC commands there, and you see on the VNC display what > would appear on the Windows VGA display. OK, Thanks. 2 other questions: 1. how do I connect tzo the VNC server for my domU OS? 2. Where is the QEmu code? In xend or in kernel? -jec Jean-Eric Cuendet Senior developer / Technical support

It looks like there is a bit of a problem with slackware distributions. I've tried this on a 3.6 and a 4.0 install with the same results. The first thing is the autoconf doesn't check for the sys/select.h file and it's not present on my systems. The autoconf for 1.2.26 does this check. The second thing I bumped into is: gcc -o sshd sshd.o auth-rhosts.o auth-passwd.o auth-rsa.o

Hola, que tal? Tendo una duda, como se puede eliminar una variable de un data.frame? Tambien tengo otra duda, tengo una base de datos importada del spss, donde tengo espacios asignados como missings, pero al pasarlo al R, me coje los espacios como una categoria más. Como los puedo poner como missings? Dolors Giralt _________________________________________________________________ [[elided

I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is

Hi All, I am looking for a provider/s of inbound DID - IAX numbers, for UK, USA, and Australia. Preferably free or low cost :-) Can anyone make a good reference? Many thanks Chris PS: I appreciate this is perhaps a little OT, please feel free to reply off list. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hola, ¿Alguien me puede indicar cómo conseguir que un gráfico se adapte al formato DIN-A4, con su correspondiente verticalidad?. ¿Hay que hacerlo "manualmente" o existe algún parámetro/función que lo controle?. Gracias. Eva [[alternative HTML version deleted]]

Hello all! I've noticed that rsync performs significantly worse than wget on slow congested wireless links (GPRS in my case). I don't have large statistics, but in my tests rsync often stalls for 3-5 minutes, while wget stalls only for several seconds and then continues download. Is there any rsync protocol features which are sensitive to packet loss and unpredictable delay? I

#Imagino que hay un camino más corto para resolver mi duda #Adjunto unos datos en csv library (epicalc) #Gráficos exploratorios #este comando no se ejecuta porque no ve las variables summ(IMC, by=escolaridad ) ##?porqué no ve las variables? #tuve que hacer esto que me parece que se podría hacer de manera más sencilla IMC <- (sanda[,c(1)]) escolaridad <-

Hola a todos/as: Tengo la siguiente duda: Construyo una lista de tamaño 2, del siguiente modo: miLista<-list() miLista[[1]]<-list(nombre="Ana", apellido1="Pérez", apellido2="Sánchez") miLista[[2]]<-list(nombre="Carlos", apellido1="Núñez", apellido2="Sierra") Le aplico la función turnoLaboratorio: miResul<-lapply(miLista,

Hi all, It seems that Polly could still speed up test-suite/SingleSource/Benchmarks/CoyoteBench/huffbench.c even without any optimization and code generation. Our evaluation show that when compiled with "clang -Xclang -load -Xclang LLVMPolly.so -mllvm -polly -mllvm -polly-optimizer=none -mllvm -polly-code-generator=none", the execution time of huffbench would reduced to 15 secs from

Estimados compañeros: tengo una duda sobre R; cuando ejecuto la prueba t, R calcula una t experimental y le calcula su probabilidad asociada y la compara con P=0.05 (valor por defecto). Mi duda es si para calcular este area, tipifica primera a z y luego entra en la tabla de z y la calcula y la presenta en pantalla. De otra manera no veo como asociar un valor de t a una probabilidad (la

I need to stream speex-encoded audio over RTP, which doesn't seem to be standardized yet, so I'm gonna roll my own code. I control both the sending and receiving sides, so I can pretty much do what I want. I want each packet to contain 20ms worth of audio (sampled at 8KHz), and I'm encoding using a constant bit rate and quality:6. Q: how do I determine how many bytes of data go into

Eva, ¿Algo asi o te entiendo mal? l <- list()l$nombre <- "Ana"l$apellido <- "Pérez"parametro <- "nombre"if (exists(paste('nombre', 'OnValidate', sep=''))) nombreOnValidate(1:5) # cuidado con el enviroment nombreOnValidateif (exists(paste('s', 'um', sep=''))) sum(1:5) # esto lo agrego como ejemplo[1] 15l[

I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't