similar to: definity prologix

Hi My name is Jose Manuel Cortes and im developing an IP telephony project, im going to interconnect a definity prologix PBX with an asterisk server (i still don't know what kind of cards i'll use digium, sangoma or voicetronix)trough a E1 connection in order to add ip telephony to the university. I would like to know if there's a compatibility problem with this. I send you a

Hi Im also new but you should know very well all the interfaces you are going to connect the sistem, the number of users you'll have (hardware requeriments), know a lot about the soft/hardphones you'll use and download the asterisk handbook or the big one (i don't remember the name) Good luck Jose Manuel Cortes David X Semestre Ingenieria Electronica PONTIFICIA UNIVERSIDAD

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

Hello, I was using version 2.13.2 and I have just downloaded the latest version 2.14.1. However, I''m trying to install the packages I was using and when I look for them in the packages list, I can´t find many in the CRAN binaries (e.g. "vegan"). I do find them in the CRAN sources but the installation fails. I tried downloading the version 2.14.0 and I had the same problem. I

Hello, I'm trying to figure out how to do something that I hope is pretty easy. I have a remote phone system (Definity ProLogix) connected to my Asterisk system via a T1 cable (all onsite). I'd like to get some of these users on a queue hosted on the Asterisk. I've got it setup so that it seems to work OK (calls flow normally), but I'd like the users to be able to dial one

Hi R-Core, i am using nnet and DEoptim, Xcc=matrix(rnorm(100,0.5,0.08),50,2) Ycr=matrix(rnorm(50,0.2,0.05),50,1) pred_regm1 <- function(A) { A1=A[1] A2=A[2] A3=A[3] regm1 <- nnet(Xcc,Ycr,entropy=T,size=A1,decay=A2,maxit=2000,trace=F,Hess=T,rang=A3,skip=T) dif=sum((predict(regm1,Xcc)-Ycr)^2) return(dif) } somar=DEoptim(pred_regm1,c(1,0.00001,0.01), c(25,0.999,0.95),

Hello forum, I want to do a test of independence with svychisq, but I get an error, then this my code: am18 <- read.spss("C:/Users/diana/Dropbox/Semestre 10/Tesis 10/Tesis Diana/AMcomuna18-29MAR2012.sav", use.value.labels=TRUE, max.value.labels=Inf, to.data.frame=TRUE) b<-matrix(c(am18$N6_MANZANA),ncol=1) c<-matrix(c(am18$PM1_1_PONDEMUESTRA),ncol=1)

Hi, I bought a Digium Quad E1 card model TE406P. Till now, I can't make it work... I mean, I have red alarm when I configure one E1. The provider is in France (France T?l?com) and I use the following zaptel config : span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 I'm using Linux 2.6.15 and when I run ztcfg -vvvv, it seems that all channels are configured... So can someone

Hola quisiera saber de que manera puedo generar numeros aleatorios enteros entre 0 y 4, para completar una matriz de un automata celular Muchas gracias Juan Nicolás Corral GómezEstudiante Carrera de EcologíaPontificia Universidad Javeriana [[alternative HTML version deleted]]

please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . --------------------------------- Yahoo! Mail Use Photomail to share photos without annoying attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hola, ¿como están? Me encuentro haciendo un trabajo de Explotación de Información para la facultad y tengo un inconveniente a la hora de preparar los datos para poder operar con ellos. Tengo un atributo "Titulo Secundario" que cuenta con 139 valores diferentes (al final les dejo algunos de estos valores con la cantidad de ocurrencias de cada uno) y lo que quiero hacer es

Gracias chicos por sus opiniones y recomendaciones. Como dice Javier Rubén, justo lo que quiero es que los chicos aprendan bien los fundamentos de un lenguaje de programación orientado a objetos para que no solo copien el script de una librería, corran y ya. Quisiera que comprendieran los beneficios de R en términos de manipulación de sus bases de datos y optimización de procesos a través de los

Hello, since I had delayed the programming of the UPS Device part, now it is time to continue... My Device is "24 V DC modular ATX PSU" which replace the standard ATX Power Supply in PCs and use 15-43 V Input Voltage. The microcontroller is a TI LM3S5T36 Cortex M3 and I get the "UPS" infos using up to 4 Maxim DS1780 and some I?C current sensors. The ATX PSU has 4

Hi all, I have a Postfix+Dovecot working in beta/testing; it seems to run very fine, but one beta-tester user says that kmail informs him about frequent desconnections. My dovecot -a shows: # /etc/dovecot.conf base_dir: /var/run/dovecot log_path: info_log_path: log_timestamp: %b %d %H:%M:%S syslog_facility: mail protocols: imaps listen: [::] ssl_listen: ssl_disable: no ssl_ca_file:

You should run the 4ESS protocol ("a" on the Definity command "change DS1 board#) as the Definity may not send Display Name for the NI-2 protocol (settings "b" or "d") Remember to make the Asterisk zapata settings consistent with the Definity. On the Definity trunk group form Page 1, change "Codeset to send Display" to "0 (zero)", also make

I've successfully integrated * to a Definity G3SI PBX via PRI. On calls from the * box to a Definity display telephone a umlaut over an I appears at the beginning of the caller name on the Definity display. Anyone seen this before? Thanks... Bud

I'm not a big fan of Line Side for Asterisk applications.... the individual channels have to be reserved (reminds me of the "old" days in telephony) and I don't like that... rather make it a tie trunk and use ARS/AAR routing in the Definity... much more general application. This also makes routing on the Asterisk a lot simpler. Easy to make the changes.... will advise if

Hi All. I connect asterisk and definity by manual at www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya. (I just only have E1, not T1 card). I see, that card work (in definity trunk status, and at asterisk == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 -- B-channel 3 successfully restarted on span 1

We have had quite a bit of success with our T100P and TE410P cards interfacing to Nortel Meridian PBXes and also to a Livingston Portmaster 3 using ESF/B8ZS and various combinations of E&M wink and ISDN PRI (usually in 5ESS mode). In the near future, I may also need to interface to a Definity via a T1. I was planning to use PRI--is that an option for you on your switch? -----Original

We have setup asterisk to handle our calls before between telco and an Avaya definity. The PRI keeps locking up every so often. In addition I keep getting this error when trying to call the avaya: -- Channel 0/2, span 1 got hangup request, cause 102 -- Hungup 'Zap/2-1' When that error happens I get a fast busy (congestion) tone. Any one can point me in the right direction? TIA