similar to: ChannelRedirect

Hi, I'm struggling with a feature in my home phone setup. I have several phones using both SIP and SCCP. What I try to do is to create a dynamic feature that works similar to the blindxfer feature built into Asterisk. What I want is the possibility for the called part to push a number sequence (for example *#) to redirect the callee to a fixed extension or (for example *123#) to redirect the

Hi, i'm trying to implement the following scenario: - A user calls number 700 - Asterisk then dials to extensions 100, 200, 300, 400 and 500 - And then bridges all calls to a conference room I tried to use MeetMe and ChannelRedirect, but seems that after channel redirect nothing more is executed. So, this seem to work for the caller and first called, but the others

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

Hello all, I've got some PHP code that opens an AMI socket and does a ConfBridgeList for a specific bridge (8888). This all works just fine but I need to filter the information displayed to only CallerIDName so I can see a complete list of names of participants. After days of googling and playing with it, I'm no closer than I was when I started. I'm not at all married to a table.

Hello, Is there a way to either store login/logout agent information in a database or at least send an email when an agent logs in or out of a queue? Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110812/84130e1a/attachment.htm>

Hello, I'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Dear R-user, I have a data like this below, age <- c("young","mid","old") married <- c("no","yes") income <- c("low","high","medium") gender <- c("female","male") I want to make some of combination data like these, age.income.dat <- expand.grid(age,

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Hi, I am using Asterisk 1.8 and trying to pack some custom data in a CEL HANGUP event. In a master (inbound) channel I can set the CHANENL(userfield) to pass custom information to a CEL event. In the outbound channel created by Dial() I can also possibly use a macro/gosub on answer and set the CHANENL(userfield) from there. The problem is how to set it in an outbound channel created by Queue()

I'm having trouble getting the quota plugin to work with deliver. The quota in the database is not getting set. I am using dovecot-1.0-rc15. This server is not yet in production. Some values below are because I'm trying to test with the simplest possible case, and some because it's reading from a different database. Also, I noticed that "dovecot -n" doesn't display any

Hi there, I am new to R and feel so bloody stupid. Abut in spite of a search of several hrs I could not find an answer to my problem. I have imported SPSS data to R, and apart from some warnings regarding duplicate labels, everything looks fine and names lists all my variables. Then I try to run a seqdef command - and get the error message below: > fam.seq<- seqdef(family_seq, var= c

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Dear R users, I have a problem, which I can not find a solution. Probably someone could help me? I have a result from my classification, like this > credit.toy [[1]] age married ownhouse income gender class 1 20-30 no no low male good 2 40-50 no yes medium female good [[2]] age married ownhouse income gender class 1 20-30 yes yes high male

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

Guys, I am using Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org on a x86_64 running Linux on 2013-01-18 19:52:25 UTC How can I set variable in one context and then Redirect a channel to another context and use variable there? The code below doesn't work, so I've got empty VAR1 in context_2 [context_1] exten => s,1,SET(__VAR1=VALUE1) exten =>

José, La función survfit espera como argumento una fórmula. Al pie del help de la función survfit dice: "Versiones anteriores del código también permiten la especificación de una sola curva al omitir el lado derecho de la fórmula, es decir, ~ 1. El manejo de este caso requiere algunas manipulaciones poco comunes y frágiles, este caso ya no es compatible." (sepan disculpar la

Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf

En el libro EPICALC (pagina 229-230) en el que está el siguiente script, todo nos funciona bien, pero cuando vamos a life table, ya allí no avanza, lo señalamos en el script, por favor quizá se nos haya ido algún detalle, pero fuimos siguiéndolo por el libro paso a paso y no no hemos percatado Todos los de el paquete survival de la ayuda del R funcionan perfectamente

Good day, Our installation needs a robust ACD application (as I'm sure others do) that can be dynamically reconfigured (if possible) maybe by a MySQL database. I have looked at Bill Heckel's ACD work and Andreas Otto's DynExtendb as well as James Sharp's ACD. None of these seem to be quite finished. Is there a chance a unified (* blessed) ACD application can be put