similar to: Running applications when a queued callisanswered

>From the queues.com file. ; An announcement may be specified which is played for the member as ; soon as they answer a call, typically to indicate to them which queue ; this call should be answered as, so that agents or members who are ; listening to more than one queue can differentiated how they should ; engage the customer ; ;announce = queue-markq This allows you to have one

Hello, I'm experimenting with Asterisk for possible use in a call center. I'm trying to figure out how to run applications when an agent answers a call in the queue. I see that the queue itself supports a very limited range of applications; for example, I can give a URL to the Queue() application to SendURL(), or an announcement to read to the agent. I'd like to do some slightly

When I do a 'asterisk -vvvvvc' I get following, but asterisk does NOT stay up: == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Hi all, I'm unable to put app_festival to work. I successfully patched, installed and tested festival (interactive logon and telnet to server port) which seems to work without problems. But when I test it in asterisk I got the following trace in console: -- Executing Answer("SIP/bsenicar-850b", "") in new stack -- Executing

Hello, Suppose a dialplan such as: exten => 6004,1,Answer exten => 6004,n,Wait(1) exten => 6004,n,SayDigits(1) exten => 6004,n,Festival(This is a test of Festival) exten => 6004,n,Hangup When watching in the CLI, I see this: == Using SIP RTP CoS mark 5 -- Executing [6004 at internal:1] Answer("SIP/505-00000004", "") in new stack -- Executing [6004 at

Hello, I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk 1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine with SIP channels without Festival. I have written following context in extension.conf: [connect-to-me] exten => s,1,Answer Exten => s,n,SayDigits(?1?) exten => s,n,Festival(hello john) exten => s,n,Hangup I use call files to

I am using Festival to synthesize some menu Interaction with a caller and am having a problem. What I am working on is a remote callback where I can remotely call in to an extension, and enter a callback number (or use the CALLERID info) and a second outbound dialing number to connect to. Things work O.K. until I set usecache=yes in festival.conf. After doing this, things run well for

Trying to config the latest Asterisk/zaptel with an Digium Wildcard and a single X100m FXO interface connected to a POTS analog line. Build and install of both work ok - I'm using Suse 8 on a dual Pentium box. I load the driver with "modprobe wctdm" and the LED on the wildcard lights up. Then I start Asterisk with "asterisk -vvvgc" and asterisk fails to start. The

recently added support (with bug) for SendURL for SIP channel causes problem with nokia phones, as I reported in http://bugs.digium.com/view.php?id=9821 it was quickly resolved, but because I can't find any RFC what it is doing/how to use it, I would like to ask here, if someone using this feature, or do you know according to what RFC this was added, please let me know. thanks PJ

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

All of the Asterisk "Say" applications have stopped working. Example: SayDigits(), SayNumber(), etc... CLI output: -- Executing SayDigits("SIP/209.247.17.5-b7901508", "12356") in new stack == Spawn extension (facloc-english, 12356, 2) exited non-zero on 'SIP/209.247.17.5-b7901508' This is driving me crazy.

Hi all -- I'm having awesome fun with Asterisk & voicepulse connect together. So cool. I'm trying to have the caller id read back to me. Do I need to do something to have this sent across in the sip.conf? Or is there something I need to do somewhere to enable the reading of this data? Thank you! Matt Here is my extensions.conf exten => _XX.,1,Answer() exten

Hello, Is it possible to convert a string to a variable? I have a set of names: > stocks [1] "ACCELL" "AHOLD" "AJAX" "ARCADIS" "BALLAST" "BAM.GROEP" [7] "BESI" "BETER.BED" "BINCKBANK.NV" "BOSKALIS" "CATE" "CROWN.V.GELDER" And a

I'm Japanese. Sorry,English is not so understood,Please let me question by items. In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of Authenticate application's 'j' option. exten => 123,1,Answer() exten => 123,2,Authenticate(789,j) exten => 123,3,Playback(pin-number-accepted) exten => 123,4,SayDigits(111) exten => 123,103,SayDigits(999) In this

> Date: Mon, 2 Feb 2009 12:56:15 +0100 > From: friedrich.leisch at stat.uni-muenchen.de > Subject: Re: [R] Problems in Recommending R > To: thomas.petzoldt at tu-dresden.de > Cc: "r-help at r-project.org" <r-help at r-project.org>, > useR-2009 at r-project.org, paul at stat.auckland.ac.nz > Message-ID: <18822.57183.637787.426445 at

Hello, I need a bit of handholding with R, specifically, with writing packages for it. I'm a systems programmer, and am, on the request of several users of our software, working on generating R interfaces. For starters, I've written the following R function (which compiles): SEXP myincr(SEXP Rinput) { // Returns input integer incremented by one int input; SEXP returner; PROTECT(Rinput =

I have the following in my extensions.conf: exten => 21,1,NoOp(${CALLERIDNUM}) exten => 21,2,GotoIf($[${CALLERIDNUM} = ""]?21|4:21|9) exten => 21,4,Playback(/etc/asterisk/interactive-services/no-callerid) exten => 21,5,Wait(1) exten => 21,6,Playback(/etc/asterisk/interactive-services/no-callerid) exten => 21,7,Wait(1) exten => 21,8,Goto(10,4) exten =>

hello, I use trixbox.I had define a feature code testfeature: [applicationmap] #include features_applicationmap_additional.conf testfeature => *3,callee,Macro,vote [featuremap] blindxfer => ## ; Blind Transfer disconnect => ** ; Disconnect Call automon => *1 ; One Touch Record atxfer => *2 ; Attended Xfer testfeature => *3 here is my macro-vote: [macro-vote] exten

Is there a list of phones (hard or soft) that support the Asterisk SendText or SendURL applications? I have been trying to make this work with a Snom 200 and a Cisco 7960, to display call progress information, such as which trunk a outbound call is routed to, but my attempts have been unsuccessful. The Snom claims to support SMS somehow, but searches reveal no useful information on how it is to