similar to: Ring a grop of extension, then playback a file, then transfer to external number

Hi all, Here is the situation. Linux workstation access a web page on a web server (not the one running Asterisk). From that web page, we need to initiate a dial-out on the Asterisk server and once the call is connected, it must ring on the agent's hard phone. Any pointers about how to initiale an Asterisk call from a remove web server? Thanks, Andre Courchesne

Hi, Anyone knows if there is a way to play a list of sound file in a round robin mode (at specific interval) while someone in waiting in moh in a queue? Ok, you enter a queue and wait listening to moh, every X minutes a sound file is played from a list of sound files to be played. If that possible and if so how? Thanks for any pointers. Andre

Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display "Confidential" or "unknown" as we sometimes see ? Andre

Hi, For thanks to everyone that answered the "dial from pph". On an other subject, how would I go about playing a wav file while talking to someone over a Zap channel ? Let me explain. I am on line with someone. I want him to hear a WAV (or mp3) sound file. I punch a key on my phone keyboard and he hears the sound file and after we can continu talking. Any hints

Prepend *67 if your carrier allows it Thanks, Steve Totaro http://www.asteriskhelpdesk.com > -----Original Message----- > From: Andre Courchesne - Consultant [mailto:courchea@net-forces.com] > Sent: Thursday, April 13, 2006 12:02 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Display "Confideltial" or "unknown" on called > iddisplay

Hi all, First, this is not my first PRI/T1 Asterisk deployement. Did several with Bell, Telus, AllStream, Rogers but this is my first with Videotron. Just spoke with the person taking the order and on top of the standard settings (switch, coding,...) she asked me about data rate (56k or 64k). Since I have never been asked this question before and can find anything relevant in the

We have 1 server that after a few hours operating has multiple process of asterisk running. Here is the pstree output: # pstree init-+-atftpd |-auditd---{auditd} |-bash---safe_opserver---op_server.pl |-crond |-cwASTcall.pl |-dbus-daemon |-events/0 |-hald-+-hald-addon-acpi | `-2*[hald-addon-stor] |-httpd---3*[httpd] |-khelper |-klogd

Greetings, I'm trying to setup an option in my greetingmenu that would allow the caller to select this particular option for emergency calls. That option would dial out on an available PSTN line to a cell phone number. Currently it is setup as such exten => 9,1,Dial(Zap/g1/<CELLPHONENUMBER> where <CELLPHONENUMBER> is the number it is calling out to. When option 9 is

I have written a function called mcsamp() that is a wrapper that runs mcmcsamp() and automatically monitors convergence and structures the inferences into vectors and arrays as appropriate. But I have run into a very little problem, which is that mcmcsamp() shortens the variable names. For example: > set.seed (1) > group <- rep (1:5,10) > a <- rnorm (5,-3,3) > y <-

Hi, Any reason why I can not get the php exec() function to execute a shell command inside an agi script? Thanks. Andre

There seems to be something about SL for queues since when the show queues CLI command is used, it give something like "SL:0.0% within 0s": pbx*CLI> show queues 1 has 3 calls (max unlimited) in 'rrmemory' strategy (243s holdtime), C:174, A:9, SL:0.0% within 0s Members: SIP/1242 (dynamic) has taken no calls yet SIP/1251 (dynamic) has taken 4 calls

Hi! I have samba 3 beta2 as PDC. Now I need to make all mebers of the unix grop "users" local admins on their Windows systems, because Wordperferct 8 doesn't run otherwise. As the "domain admin group" setting from smb.conf doen't exist anymore, I don't know, how to do the group mapping correctly. Could someone explain the steps to do it? Thanks in advance for

Hi, Is there a way to create trunk groups while asterisk is running. For exemple let's say that zapata.conf defines g0 as channels 1-23 I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23 Any hints appreciated. Andre Courchesne

Hi, Anyone has good/bad experience with SIP providers in upstate NY? Any recommendations of such provider who works great with Asterisk? Thanks, Andre Courchesne

Hi, I have a problem where some PRI channels get stuck in a "Call" mode. If I do a zap show channel XX, it shows as "PRI Flags: Call". However there is no calls on that channel. Trying to force a hangup does not work: [root@neil1 Dialer]# asterisk -r -x "soft hangup zap/27-1" -- Remote UNIX connection zap/27-1 is not a known channel Any ideas?

Hi, Ok, here is the setup. Asterisk conected to a PRI line (23 lines). 3 tool-free phone numbers are routed to this PRI line. Customer wants to have a way to have shown on the receptionist phone that the call comes from which of the 3 tool-free lines. Possibly display on the phone that the call comed from tool-free number 1, 2 ou 3 or even better a name or text id associated with this

I've been trying to keep track with lmer, and now I have a couple of questions with the latest version of Matrix (0.995-4). I fit 2 very similar models, and the results are severely rounded in one case and rounded not at all in the other. > y <- 1:10 > group <- rep (c(1,2), c(5,5)) > M1 <- lmer (y ~ 1 + (1 | group)) > coef(M1) $group (Intercept) 1 3.1 2

Dear r-helpers, I can run the examples on the mcsamp help page. For example: **************************************** > M1 <- lmer (y1 ~ x + (1|group)) > (M1.sim <- mcsamp (M1)) fit using lmer, 3 chains, each with 1000 iterations (first 500 discarded) n.sims = 1500 iterations saved mean sd 2.5% 25% 50% 75% 97.5% Rhat n.eff beta.(Intercept)

Hi, It's my first attempt to run asterisk 1.4 (have been on 1.2 for a while) and I have a problem where playback and background are played very very fast. When I say fast is you get a few sounds that's it... Running kernel 2.6.20.4 and latest released asterisk packages (asterisk, libpri, zaptel). Any hints? Andre COurchesne

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................