similar to: Digium X100P card to a brazilian analog line

Folks, I'm fairly new to this whole Linux UPS thingie, but I'd quite like to have a look at getting my UPS to work under Linux and would be grateful for any help in getting a driver. I have a reasonable working knowledge of Linux and software development, and thus am happy to modify config files, alter kernel settings, etc, although I'm no C guru. I have a Kebo UPS-650D, which

Dear Xen community, I have Xen 3.0 installed on RedHat Linux Enterprise RHEL4U2. "xend install" runs fine with no error messages. However, when I start "xm cr guest-vmx.conf" I do not get any new window open for the new guest OS. "xm list" shows that the vmx has started and seems to be working fine (just for testing, when I type "xterm" an X window

Hi All: I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling. Thanks Begra8fl >From: asterisk-users-request@lists.digium.com >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs >Date: Tue, 04 May 2004 13:32:00 -0500 > >Send

Hi all, I need help. I have a Linux box with SER as a proxy server with ip phones attached on it , and another linux box with Asterisk and T410 card connect to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk and then to SER box and then to the phone .every time an invalid number dialed from PSTN to SIP phones connected to SER asterisk says that the call is progressing

Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

Happy holidays! I am experiencing an issue when trying to check my mail using IMAP. with Dovecot I have tried checking my mail using a full GUI client (Thunderbird) and telnet. Both times I get disconnected before all of my messages can be downloaded and I see an error in my mail log. Here are the details: [root at cust19-1-prod-domain userqa]# dovecot --version 2.0.9 [root at

Hi, I tried to install R-2.4.1 on cygwin system. "./configure" succeeded, but make failed. Below, I provide the output from the process: error message, and info from configure output, in that order. I appreciate that someone can guide me (technically in-sophisticated) through this process. Again, thanks for your help. Michael Niu (1). Output from make make[3]:

=20 =20 12:53:41.358166 IP (tos 0xb8, ttl 127, id 0, offset 0, flags [none], = proto: UDP (17), length: 856) 189.8.113.170.5060 > 189.8.126.177.5060: SIP, = length: 828 INVITE sip:7002 at 189.8.126.177:5060;user=3Dphone SIP/2.0 Via: SIP/2.0/UDP 189.8.113.170:5060;branch=3Dz9hG4bKba4h2m2070fhnc4q20k1.1 Call-ID: d6dc25017b171144f35fb9e1c9c393a3 at 10.0.0.10

and Asterisk is plugging in pseudo ID. Is that correct? It seems to me that Asterisk should simply say "no caller ID" or "No ID" or something besides "Asterisk". In any case, we are trying to filter them with little success. When we do a LEN(CALLERID(num) we get "13", when we expect "10" The call pattern is 1 call followed by a

community there is a real possibility this may come off so if you have an interest in this space and want to contribute to the discussion then this is your opportunity to do so. =20 I look forward to all opnions on this topic. =20 The slide deck for the agenda of this call is located here http://voipusersconference.org/2008-05-09-Slides=20 Cheers, Dean=20 ________________________________

using Asterisk. =20 Is this all you want Asterisk to do? (eg as an application service rather than provide telephony for the rest of the library as well), or are you looking to have it replace your existing telephony equipment? =20 As a suggestion if you google Trixbox and Nerd Vittles you will find a fairly detailed explanation of how to set your Trixbox server (a version of Asterisk) up to

display, accelerometer/motion sensor being the first 4 for release (though 81 have been mocked up so far). The long term concept is if you want a 'weather station with live video feeds and gps location control you can add various modules together to deliver what you are looking to achieve. I have high hopes for the concepts, and wish the guys well as it seems their hearts are in the right

1. a clause in iphone Developpers agreement that forbid applications runnin= g in background, 2. lack of sip clients. Now it seems skype is available on iphones. Has someone tried it ? Along new skype capabilities in Asterisk, could it be used to hook iphones = to Asterisk for both inbound and outbound calls ? Regards --_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0242cworksmailcwo_

using Asterisk. =20 Is this all you want Asterisk to do? (eg as an application service rather than provide telephony for the rest of the library as well), or are you looking to have it replace your existing telephony equipment? =20 As a suggestion if you google Trixbox and Nerd Vittles you will find a fairly detailed explanation of how to set your Trixbox server (a version of Asterisk) up to

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

don't have a public facing web page but you are looking for people to click on but a personalized list of numbers. In order for someone to access this directory you are going to be asking for a username/password correct? If so just tie the username to a selection of 'my location' checkboxes that I tick and then the app remembers this location next time I log in (eg server side

"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know= your iPhone." --_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0248cworksmailcwo_ Content-Type: text/html; charset="us-ascii"

might be your best bet to get the information you want. I'd look at voip-info.org for information. _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier Sent: Wednesday, September 16, 2009 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to list ongoing calls

_____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of khalid touati Sent: Tuesday, April 13, 2010 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Time variables in system application Hi Guys, i have a weird thing here: when using time variables (%F & %T) in a shell script, out