similar to: Confference function

Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Caller should hear the called party but not vice-versa) - A DTMF sequence (maybe handled in features.conf) for

Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently

Hello, When dialling a SIP phone which is already in a call the caller hears a "regular" ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi:

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use ChanSpy or

Hello all, I currently have all outgoing calls set to mask the caller id so it will always appear to be coming from our main number. The problem I'm having though, is with both the call detail in mysql and with the automon (recording) feature. It shows the originating number as the number I masked it to, rather than the actual person calling. How can I go about having both the destination see

I am looking for a way to automatically close a meetme conference when either a user hangs up or through an agi call? Some method that would automatically terminate the meetme. Is there a way to do that? Jerry

All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. Any suggestions? - sf

Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, as I wrote in a previous thread I was experiencing dropped audio (apparently randomly) and SIP + IAX peers getting REACHABLE / UNREACHABLE without reason, servers were in the same LAN. Investingating deeply in the problem I also noticed that 'show channels' command on the CLI, sometimes were returning strange results, for example it wasn0t showing some channels I was sure

hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because

hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq --

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use

Dear R users, I would need some help in building R using minGW in windows 8.1. After using the command *configure *(./configure --enable-R-shlib --with-readline=no --with-x=no), I use the command *make. *This results in the following error: [...] make[2]: Leaving directory `/home/Edoardo/r-3.1.2/src/nmath' make[2]: Entering directory `/home/Edoardo/r-3.1.2/src/unix' make[3]: Entering

Hello everyone, I'm in the process of running an Icecast server and I would like to know some best pratices. 1. Should I place Icecast on port 8000 or should I change that to one more common (80, 443...)? 2. Should I place the server behind a webserver like ngingx or apache? 3.Can I disable the login interface? what can be disabled? My best guess is to run icecast behind a webserver,

Hi all, I have a problem with Asterisk dropping audio. While in call, audio gets dropped for a while (from 5 to 60 secs, and obviously users often hangup, this means that I'm not sure the audio is always coming back after 60 secs), in the meantime the call remains up and no SIP signalation is generated. It happens randomly so it's very difficult to debug. I cannot see common

Hi, I'm trying to use reshape but I cannot quite understand how it works. Could somebody help me on this? Example, my data is something like: mydat <- data.frame(tree= 1:10, serra=rep(1:2, c(5,5)), bt01= 101:110, bt02= 201:210, bt03= 301:310, mm01= 9101:9110, mm02= 9201:9210, mm03= 9301:9310) > mydat tree serra bt01 bt02 bt03 mm01 mm02 mm03 1 1 1 101 201 301 9101 9201

How can I get a list of all model classes in the domain of a Rails application (all models, both in "app/models" and in components/somedir/model.rb)? Thanx in advance for your precious help! Edoardo "Dado" Marcora

Hello erverybody, I have a problem with my second for-loop. 1. First i read the tables. datos.mx1 <- read.table('PETmx1.csv',head=TRUE,sep=';') datos.min <- read.table('PETmin.csv',head=TRUE,sep=';') http://r.789695.n4.nabble.com/file/n4638257/PETmx1.csv PETmx1.csv http://r.789695.n4.nabble.com/file/n4638257/PETmin.csv PETmin.csv names(datos.mx1)

oops, I hit the return and it got sent. All those questions depends on the application you setting up. if your worried about security, isolate the icecast server from your network. We designed a user interface on the company web site fronts the stream servers. The customers get a simple URL for the stream. We use several stream servers some on site and some off, our web server setup to server