similar to: Loopback

I've got a server running FreeBSD 6.2 and PF. The server has a couple dozen jails on it. Previously, I had a few "private" services such as MySQL running on loopback IPs (127.0.0.2+) and the rest of the jails running on the public IPs. I have to renumber my machine with a new block of public IPs so I thought I'd be clever and move all the jails onto loopback IPs. Then

Hi all, I want to initiate a call from the asterisk to an extension, where I will forward the asterisk side to another extension later (to the conference extension). I can initiate a call uning originate call from an extension to the desired extension, but it would need someone from the originator extension to answer the phone. How can i register an extension to asterisk where it

Hi all, I've note that the variable DIALEDPEERNUMBER is broken. Now i want to know if exist another method to retreive the called number on *, and, if it's possible, an example ;) Regards.

Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND

Hello, After battling with the Asterisk Manager interface(and getting it to pretty much do everything I want to do with it) I thought I'd share my experiences with those who are developing or are thinking of developing applications using it. First here's a list of some of the things the manager interface will let you do: - Dial a call from any extension/resource to any other

check the following code: # settings n <- 100 # number of sample units p <- 10 # number of repeated measurements N <- n * p # total number of measurements t.max <- 3 # parameter values betas <- c(0.5, 0.4, -0.5, -0.8) # fixed effects (check also 'X' below) sigma.b <- 2 # random effects variance # id, treatment & time id <- rep(1:n, each = p) treat <- rep(0:1,

Hi, Running: Freebsd 6.0 I am wondering if it is possible to have acces to loopback ip in a jail. I currently have a server running a jail. In the jail, there is a database and a web server. I would like to be able to have the database only bind on a loopback address and not on the jail's ip. Can this be done and how? Thanks -Cyril

Hi, I'm having some problem with a dataset and I don't really know how to analyse it. I have 20 subjects in two groups of treatment (8 an 12 subjects). Biological measure have been recorded at different time, from t0 (before the treatment) to t7 (3 days after). The time elapsed between each measure is not constant. What is the most appropriate test to show a difference between the 2

> I recently did something like this. I have a webserver in a jail that > needs to talk to a database, and the webserver is the only thing that > should talk to the databse. > My solution was to use 2 jails: one for the webserver, and another for the > database. > Jail 1: > * runs webserver > * binds to real interface with real, routable IP > Jail 2: > *

Hi All, I use the Asterisk Manager Interface to redirect the channels. There have two channels : SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456) SIP/612-5456 s@macro-monitor:10 Up Dial(SIP/0882@voip_out Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel:

Me ha parecido una opción muy interesante la web roncloud.com comentada por Carlos Ortega. ¿Alguien sabe si pueden subirse ficheros R.data a esta web? Muchas gracias Pilar de la Cruz Message: 4 Date: Fri, 28 Jun 2013 22:33:09 +0200 From: Carlos Ortega < cof@qualityexcellence.es> To: Lista R <r-help-es@r-project.org> Subject: [R-es] R en tu navegador... Message-ID:

Dear friends I need to loopback the audio on my channel. Did anybody on the development team thought about a function or app that would do that? If it is not clear, I mean that whatever audio I get, I send back. Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130520/e50219b2/attachment.htm>

Hi! I'm trying to get t38modem working and started with the loopback mode. I installed everything according to http://www.voip-info.org/wiki/view/T38modem+configuration+with+Asterisk and used a stock asterisk 1.4.20.1 compiled from source. Openh323, ptlib_unix and t38modem are compiled from source as well, as outlined on the wiki. Hylafax starts sending the job (sudo faxstat -s shows one

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

On Fri, Dec 5, 2014 at 3:23 PM, Rodrigo Montiel <guevara2309 at yahoo.com.ar> wrote: > Hi masters, > > I?m not an expert on this my friends, but I?m trying to understand which the > expected behaviour is from Asterisk side when you deal with the following > scenario: > > Caller ?> GSM Gateway with SIM card A ?> Asterisk queue ?> extension 1000 > > GSM

I have the following issue: 1) Call comes in from PSTN to Asterisk (IP A) and Asterisk forwards call to a SIP Proxy (IP B) 2) SIP Proxy (SER) forwards the call to a registered user. User does not answer and Call Forwarding is turned on for the user and the number to forward the call is a PSTN number. 3) After a specific timeout, SER has to forward the call to the "forwarded" PSTN

I'm very interested in the redirect feature of Asterisk. So far I haven't gotten it to work. My scenario is that there is a two party call going on where I want to send one of those parties somewhere else. In the wiki is only an example how to send both parties to a meetme room. Is the ExtraChannel parameter required? This is what I have: Action: Redirect Channel: SIP/8080-e2a7

Hi! I'm building an application which allows to dial via the Asterisk Manager Interface using the originate command. There should be an optional conferencing feature. The manager commands are basically: --------------------------------- action: login username: sdjklgdsjg secret: xxx events: on action: originate callerid: 3847438609 priority: 1 exten: 4068439865 async: 1 context: out

If you have two asterisk systems how do you hook them up together so the users of one system can make calls onto the other system. Thanks Steve steve@17q.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040927/69058745/attachment.htm

Can the Manager Redirect command transfer a ringing SIP extension? I'm trying to implement a Camp On feature, and having failed to do it in Dial Plan logic, am trying to do it with manager logic. If an arbitrary Sip extension is ringing, I need the ability to pick up that extension from any other phone. What little docs there are on Manager commands shows Redirect takes these parameters: