similar to: usb phones in linux, any??

Hi all, i''m new at LARTC, and after reading the docs I found no solution to my problem ... On one side I have eth0 conected to the LAN, on the other side I have eth1 conected to a switch and to 3 DSL routers with 3 diferent providers, and also eth2 conected to a cisco 2600 conected to a LDMS line. I have readed the larct docs about multihomed conections to internet, but I''m

This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of

I found this information on how to make XP have a dialpad in Windows Messenger which was awesome news HKEY_CURRENT_USER\Software\Microsoft\MessengerService\CorpPC2PHone (change it from 0 to 1 and a magic new choice to make phone calls appears) only to be crushed hours later when I realized It doesnt seem to do dtmf right. If i make an ext lead to AgentLogin for instance and press my

Hello, I'm trying to understand why when I make a call from a SIP phone to an external number who has an IVR system in which I've to choose some options using the dialpad, it does'nt recognise the key pressed and remains still waiting for my choose. I'm tryng using Grandstream 102, and i've tryed with all the 3 modes possibile: Dtmf inband, rfc2833 and INFO (obviously

I'm using happily this cheap phones, but I still have a little problem. Configuring the phone is extremely easy on * and I've a couple of them perfectly working, except when i try to call some toll-free number (in italy 800xxxxxxx ). If the number called is an IVR system, often with GrandStream (but also with Cisco 7905.h323) it's impossible to make the menu choices via the Dialpad.

Greetings, I need a 2 way intercom for separate kitchens to communicate without having to walk back and forth. The speaker has to be loud but clear, not distorted. Sometimes the kitchens can be noisy. It needs to be easy to use. It needs to be easy to clean. It would be nice if it used POE. Eventually I would like the kitchens to be able to dial different parts of the restaurant when I

Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan,

All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is

Well I think we all need to look at something like this first. We will be one of the first people in Europe who will be selling this. If anyone is interested do drop me an email. Picture of the phone can be found here. http://cyber-telecom.net/wifi-gsm.jpg GSM / VoIP Over WiFi Dual-Mode Phone CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi dual-mode smart phone, in

Before I relate the actual problem, some context. Callcentre environment, a few users testing a new digital dialer... 1. Agents are using Grandstream ATA HT486 and a small analogue dialpad with a headset. 2. SIP connection to Asterisk-1.2b1 3. IAX2 connection to ITSP provider. The call is initially set up in the following way. 1. Agent calls into a meetme conference room and subseqently stays

Hello everyone: I have a problem with the shorewall configuration. Let''s me tell you. I have installed shorewall 2.0.4 into a machine with 2.6.8 kernel. This machine works like a software-router: it has 2 netcard eth0 goes to the local network 192.168.0.0/24 eth1 is an interface for ppp0 (there is an ADSL conected) I have defined the Network Zones (net, loc); The Network Interfaces

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

I wish to make modifications to the plot.pedigree function in the kinship package. My attempts to contact the maintainer have been unsuccessful, but my question is general, so specifics of the kinship package might not be an issue. My first attempt was to make a new function Plot.pedigree in the .GlobalEnv which mostly achieved what I wanted to. However, I'm sure that's not the tidiest

I like the fact that the postscript function enables the possbiility of a more useful title than before. However, I'd prefer the default to be the file name. It's very simple for me to make my own postscript function that does just that simply by setting title = file. I always use onefile = TRUE, so it always works (so far). However, I'm a little reluctant to do that in case some

I like the fact that the postscript function enables the possbiility of a more useful title than before. However, I'd prefer the default to be the file name. It's very simple for me to make my own postscript function that does just that simply by setting title = file. I always use onefile = TRUE, so it always works (so far). However, I'm a little reluctant to do that in case some

Somone in europe have had succes getting Callir ID showed on a phone screen conected to an Handytone 286 ? Adri? Vidal -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 235 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/e5514052/attachment.bin

Dear R users, I have a problem in the configuration of R: I just changed university, and my conection to the net is via a password, which permits me to access the packages with no problem via the internet explorer (version 6). I just updated R to R 1.81, and I cannot download nether upgrade packages (and many are not working with the update!!!). I have been looking in the help and emails, and

I have a E1 conected to asterisk all zap channels are ok, but when calls come into Asterisk caller don't hear none ring, the call goes straight into the menu, how can i simulate 2 or 3 rings? here it is my conf. exten => s,1,Answer exten => s,2,Wait,2 exten => s,3,NoOp(${CALLERID}) exten => s,4,ResponseTimeout,45 exten => s,5,DigitTimeout,3 exten =>

Signif. codes: 0 *** 0.001 ** 0.01 * 0.05 . 0.1 1 Signif. codes: 0 <80><98>***<80><99> 0.001 <80><98>**<80><99> 0.01 <80><98>*<80> <99> 0.05 <80><98>.<80><99> 0.1 <80><98> <80><99> 1 Signif. codes: 0 *** 0.001 ** 0.01 * 0.05 . 0.1 1 hmm... they go away when I

Hi Users NUT, I want monitor a UPS of brand SMS (Sinus Double 8 KVA) using a raspberry-pi. In compatibility list, is listed to use the blazer_ser driver. I use a USB adapter to RS-232 conected in to the No-Breake. Follow the comands e confs. root at rasp:/home/pi# lsusb *Bus 001 Device 004: ID 067b:2303 Prolific Technology, Inc. PL2303 Serial Port* Bus 001 Device 003: ID 0424:ec00 Standard