similar to: Play wav while in connection with a caller

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

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, 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

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

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

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

I want to be able to send a prerecorded message to the person I am calling. I know that you can use the application map to do this. Just to test I enabled the testfeature example that is in the features.conf file. When I hit #9 during a call the other user does not hear the monkeys, they only hear a series of beeps. I have tried with different soundfiles and they all give the same problem.

Hi - I'm trying to implement the applicationmap stuff in features.conf, and I can't seem to get it to work. I'm testing it out on 1.2.2 with Polycom IP500s and Snom190s. My features.conf looks like this: [general] parkext => 700 parkpos => 701-720 context => parkedcalls parkingtime => 240 transferdigittimeout => 2 ;courtesytone = beep

I have got call recording working on our 1.4.30 asterisk box together with a recording pause ability and being able to play different audio to each party at the start and end of the pause. This all works perfectly but one wish is to have the audio files have a beep or something in them so when you listen later you can tell where the audio was paused. So I changed things around so that instead

I have a customer who would like to play a series of sound files during a phone call on demand. There would be several played in order during a call. Any simple ideas on doing that without developing a whole web app to do it via AMI? -- Carlos Alvarez TelEvolve 602-889-3003

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

I'm trying to actually use the example application map in features.conf: testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play ;tt-monkeys to the opposite channel I see the feature get registered at the CLI: == Registered Feature 'monkey' == Mapping Feature 'monkey' to app

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

Problem resolved. This makes it nice and simple to 'flash' an incoming POTS line (ZAP channel) as opposed to the dialplan scripts that I have seen that require tranferring the call, hanging up, and waiting for a call back. That was too confusing for my wife. Now all she has to do is pres *3 and it is done. No transfers. No hanging up. No dial back. extensions.conf [context] exten =>

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten =>

Hi, I am new to asterisk configuration. I want to get called number in features.conf. I am defining a feature in features.conf and that feature got executed on pressing a particular DTMF key sequence. As I want to execute my own application on pressing that key which will use called number. testfeature => 3,peer,AGI,StoreNumber|<CalledNumber> Here I want to use called number in place

Hi I'm using asterisk 1.8 on CentOS 5 I'm initiating call recordings with MixMonitor and trying to pause them with the features.conf. Whenever I try to pause the recording the call dies. Is PauseMonitor incompatible with MixMonitor? Here are some key log excerpts features reload == Parsing '/etc/asterisk/features.conf': == Found == Registered Feature

Hello, still upgrading to .20.1 and i have another issue that puppet does not regonize anymore my definitions, i use some ensure lines definition in a function.pp file and since then i am unable to make it work. puppetmasterd tells me this: Unknown function sudoensure_line in file /etc/puppet/manifests/./classes/backup.pp the import of the function is my 2 line of the site.pp and i