similar to: Vega 50 10 FXO

I am trying to configure a VegaStream 50 FXO to work with asterisk. The problem that I am having is that the VegaStream does not support incoming registration from asterisk. VegaStream only allows outbound registration. My question is does asterisk allow incoming registration from an FXO? If yes how? Or better yet, has anybody been able to make the VegaStream FXO work with asterisk? According

Polycom 601 with Sip 2.01 Anyone using Sip 2.01? I have upgraded my phones and now presence no longer functions. Buddy list shows all phones online but status does not change when someone is on a call. Also blf does not function. I am using trixbox, 1.67 was working fine on the same box. Any ideas?

Hi, dudes I got a question here. When I run "rake db:migrate" in 3rdrail, it throws an error: D:\ruby\bin\ruby.exe: no Ruby script found in input (LoadError). I have set the ruby interpreter paths, And can''t find the solution on Google. Any good ideas? Issac --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the

I have a couple of devices that need an analog modem to communicate outside of our Asterisk system. Most FXS gateways don't seem to support this... I have a stack of Sipura 2002's that are, AFAIK, worthless for this purpose. I've heard that Digium's IAXy FXS will work with modems, but I can't find any reference to that in their documentation. There is also the

I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are

Hello r-help, As far as I've seen, there is no function in R dedicated to sampling from a discrete distribution with a specified mass function. The standard library doesn't come with anything called rdiscrete or rpmf, and I can't find any such thing on the cheat sheet or in the Probability Distributions chapter of _An Introduction to R_. Googling also didn't bring back anything.

Hi, every one I'd like to introduce some new feature of our products. mg3000-r fxo gateway provides more feature to work with asterisk. 1.play asterisk ivr with no interuption. when the mg3000-r received call from co line, it wouldn't conect instantly.instead, it start call to asterisk ivr first,when the ivr ready, it connect the co line. this feature make user feel friendly.

Dear all, Which options should I use when running the configure script and/or which changes should I make to Makeconf to build the binary for R-1.9.0 but using only statically linked libraries ? I need to build a binary that is portable between several different Linux (RedHat) distributions. Thanks in advance for your help, Melanie -- http://recherche-en-danger.apinc.org

Hi, I'm looking for an ISDN FXS for home use (so the solution has to be affordable :) Let me tell you exactly what I want to do first. I want to connect a regular home ISDN phonesystem (does not exist yet so I'm flexible with that, too) to the ISDN-PSTN and Asterisk at the same time. I want to be able to place calls through the ISDN-PSTN as well as through asterisk eg by dialing 0XXXXXX

Hi all, We have several sipura 3000's working well for outbound calls, however the issue we have is that when calls are sent to the Sipura with Dial(SIP/${EXTEN:0}@sipura1) the Sipura does a SIP answer immediately and then proceeds with the call "in band" therefore sending dialing sounds back to the caller. Other SIP gateways we have notably the Vegastream and others do not do a SIP

I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep

We recently had a customer install that went horribly wrong. Serious echo (pots lines into a cac cb) that, although * did a good job getting rid of alot of it, could not get rid of it all. We tried everything, every canceller, gain setting, etc... combination possible to no avail. Both the vegastream and mediatrix boxes also could not get rid of all of the echo. So, on an off chance, we

Hi all, There is something special I must configure in order to get the voice mssage by mail? In voicemail.conf I have: serveremail=asterisk@mydomain.ro attach=yes [default] 301 => 6535,Home Mailbox,dtoma@fx.ro I have tried to let a message for 301, but this message is not forwarded by mail. I am missing something? Thanks, Dan

We have a Vegastream 400 connected to a digium Quad PRI card in an asterisk server, for the T.38 faxing here. Problem is that there are too many clockslips on it (and they get logged by asterisk as HDLC aborts). I've double checked the configuration on both sides, replaced the cable, tried different ports etc. It all lead to no resolution for it. Is there somebody on the list who has a

Does someone here know of a way to display images in R that isn't slow? Here are the things I've tried so far: library(rimage) i<-read.jpeg('foo.jpg') plot(i) # very slow library(pixmap) i <- read.pnm('foo.pnm') plot(i) # also slow chans <- getChannels(i) image(chans[,,1]) # this is slow too I also tried using library(cairoDevice) but it only makes everything

Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

I wonder if anyone can help me? Am trying to terminate to H323 Vegastream. I'm using OH323 with little success. I can dial out and answer but voip end just keepings ringing and ringing. Thanks for any help. Neil Config file: [general] listenAddress=ALL listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no jitterMax=100

Hi, We recently tried installing Asterisk for a small office. We figured the safest way to go would be to buy from someone who sold equipment specifically for Asterisk and to use a consultant that they recommended. However ... it didn't turn out so great. Sound quality is terrible -- the echo is pretty bad, and there are popping noises, too. Callers say that people on the Asterisk end

Having ISDN FXOs and a bunch of ISDN DECT base stations, I went looking for a CAPI compatible active BRI adapter capable of NT mode. I have found that the 'Eicon Diva Server 4BRI' fulfills theses requirements. However, at EUR 1387 for four BRI interfaces, it does not come cheap. Are there other users of similar hardware that can point me to alternatives ?