similar to: <SOLVED> Need help with Dial M option and destinationcontext

I would appreciate hearing from anyone who has figured this one out. Here's the scenario: I have a context wherein I give the called party the option to dial the digit 9. If he does so, he is transferred a la this extension entry: exten => 9,1,Playback(pls-hold-while-try) exten => 9,n,Noop(Attempting to bridge to ${agentext}) exten =>

I'm working on an IVR that gives the users the option (number 5 in the main menu) to dial by extension: exten => 5,1,Set(TIMEOUT(digit)=5) ; Dial Extension exten => 5,2,Set(TIMEOUT(response)=10) exten => 5,3,Background(LCL/prompt-60) exten => 5,4,WaitExten(15) When going option 5 you can dial some extensions such as 2802, it goes to the extension (all extens start with 28 on the

2008/12/2 Philipp Kempgen <philipp.kempgen at amooma.de> > Philipp Kempgen schrieb: > > Olivier schrieb: > > > >> How can you use Dial application M(x) option from extensions.ael ? > >> (As a reminder, this M(x) executes macro x when Dial called party > answers). > >> > >> It seems to me that asterisk keeps looking for this macro in >

>From my extensions.conf: exten => 888,1,Answer() exten => 888,n,WaitExten(20|m) exten => 888,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) This should: * answer * wait 20 seconds for an extension with music on the background * pass the call to that extension on ${SERADDRESS} What am I doing wrong here? I don't even get the background music while WaitExten is active. I doubt that

Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

SIP user => Asterisk 1.6 server => SIP Trunk => external destination: works AMI script => Asterisk 1.6 server => SIP Trunk => external destination: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@(ipaddr)>;tag=alphanumeric' I?ve tried doing things like ?include => contextwithtrunk" in various places, googling, re-reading relevant

exten => s,1,Answer() exten => s,n,Background(astcc-please-enter-your) exten => s,n,Background(zip-code) exten => s,n,WaitExten(5) exten => s,n,Read(NUMBER,,5) exten => s,n,SayDigits(${NUMBER}) exten => 22042,n,Dial(SIP/sipvendor/11111111111,120,A(ginger3)) exten => 22601,n,Dial(SIP/sipvendor/11111111111,120,A(ginger3)) ; xxxxx/ winchester exten =>

Hi list I encountered problem in using Background command. Below is my extensions.conf. [mainmenu] exten => 4,1,Wait(1) exten => 4,2,Background(thank-you-for-calling) exten => 4,3,Goto(n01|s|1) [n01] exten => s,1,NoOp(${CONTEXT}) exten => s,2,Background(thank-you-cooperation|m) exten => s,3,WaitExten() exten => s,4,Playback(digits/pound) exten => 1,1,Playback(digits/1)

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Every so often, and it seems that it happens only when a call is in progress, all 24 Zap channels get reset. All channels are opened and then timeout. This causes the in-progress calls to terminate. There are no corresponding Red/Yellow alarms on wither the PBX or Asterisk although we do receive a fair amount of Blue Alarms. The Asterisk server is connected to a legacy PBX through a Digium

I vote to 'fortunize' Doug Bates on Hierarchical data sets: which software to use? "The widespread use of spreadsheets or SPSS data sets or SAS data sets which encourage the "single table with a gargantuan number of columns, most of which are missing data in most cases" approach to organization of longitudinal data is regrettable."

hi, i have configured a dial plan in which i need that when a call comes to extention user should hear "main-menu" . version of asterisk is, Asterisk 1.6.0.17 my dial plan /extensions.conf [internal] exten => 2000,1,Dial(Sip/2000) exten => 2000,2,Answer() exten => 2000,3,Background(main-menu) exten => 2000,4,WaitExten(5) the problem is that when call comes it

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Hello list, I've got a problem (maybe only a problem of understanding how * works) with IAX and WaitExten. To simplify the problem I've brought it down to the following scenario: - 3 Asterisk Server A,B and C (central). - A and B both register with C. Now I want to be able to dial an extension at A to become connected to C and there I want to dial an extension to become connected to B.

I'm a bit confused about how WaitExten works. I assumed that when it returns 0, the next priority in the extension would be executed, but that doesn't seem to be the case. When I get to WaitExten and enter extension 8, it plays the message, then Waits another 10 seconds and times out. [local] exten => s,1,Wait,1 ; Wait a second, just for fun exten =>

Using dovecot-0.99.13-4.FC2 I grabbed a bunch of Mailman archive files from the Weather mailing list here: <http://www.buoy.com/pipermail/weather/> I gunzipped the older ones, then concatenated them all in correct order to create one big mbox file. I dropped it into my ~/mail folder and started my IMAP client (Mulberry) and opened the new Weather folder. It shows one gargantuan

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

All: Some of the people who dial into to our system will press the pound key when entering an extension for the directory key. When waitexten gets that, I get an error messages as, for example 123# doesn't match any extension. I was going to use ${EXTEN} to just use the first three numbers, but I'm not sure how to use this with WaitExten. so I have exten =>

Hi all, I am trying to understand how I can get a simple IVR scenario to work properly (having already removed most of my hair...). The basic requirement is as follows: * Caller arrives at our main number * Caller is greeted and then told they can enter an extension number, if known, or wait and their call will be connected to an available rep. * The IVR then dials a group of extensions (if

hi, i have a pronlem connecting call from pstn with the following confuguration, please advice extensions.conf [from-trunk] include => from-pstn [from-pstn] include => from-pstn-custom include => ext-did include => from-pstn-timecheck exten => fax,1,Goto(ext-fax,in_fax,1) extensions_custom.conf [from-pstn-custom] exten => s,1,Answer exten =>