similar to: playing "invalid" to an internal user

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

When the call comes from outside on a certain context to play "i" invalid extension to an external user is easy just by enclosing in an incoming context: exten => i,1,Answer exten => i,2,Playback(pbx-invalid) How to play an this context to an internal user, internal user has access to all contexts. In sip.conf I have [phone number] type=friend context=internal ... in

Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using

ser forwards a sip message with extension 99999996 to asterisk which plays my 'userisoffline' message and hangs up and should stop here but instead asterisk continues to process the match everything extension ._ and dials out which is not what I want... if I change the starting priority of the Dial app to a higher level than 3 asterisk stops after the hangup but then doesn't accept

When I use DISA I get congestion when I try to reach 1-800-number: Here is the context: [disa] exten => 087,1,Answer exten => 087,2,DigitTimeout,8 exten => 087,3,ResponseTimeout,20 exten => 087,4,Authenticate(985) exten => 087,5,DISA(951|disa-access) [disa-access] include => tollfree include => outgoing-voipjet [tollfree] ; ; terminate toll-free no.'s via fwdnet ; US

Dear List members- I am trying to configure ASTCC (Asterisk calling card application) but having a hard time to configure it properly. My project deadline is approaching and couldn't figure out how to make ASTCC functional. Here are some details what I have done so far. 1) I have installed ASTCC successfully. 2) I can access astcc-admin.cgi script without any problem. 3) I have created

Hi, I'm bringing this discussion here from http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ about how to manage stopping a playback on a extension previously launched with AsyncAGI and redirecting the call to another exension. If I make the Redirect without a playback, the Redirect works: http://docs.google.com/Doc?id=ahfnfrcrh3rr_30f7fzq4hd But if I make the

I tried your extension definition as suggested: exten => *98,1,Verbose(0,${CHANNEL(peername)} calling voicemail) same => n,VoicemailMain(${CHANNEL(peername)}@VoiceMail,s) same => n,Hangup But there was no change in the prompts asked, ie. the voice first asked for 'mailbox', and then 'password' as before. The prompts are not removed. Please clarify what you mean by the

Hello again dear list. Could you please help with this? Thank you for all support, you are great, and i am now at a late stage in the setup and tweaking this server, So I hope you can help me again. I Can't make include the context nighttime. Just to demonstrate if it works, I have a playback function there. But CLI reports: CLI [Jun 18 14:20:22] WARNING[2287]: pbx.c:9542

Asterisk 1.4 r181990 given the dialplan snippet below, can anyone tell me why the h exten is not being run ? ============================================================================ console output: [Mar 29 10:33:49] -- Executing [s at questionnaire-menu:1] Set("Zap/1-1", "TIMEOUT(digit)=3") in new stack [Mar 29 10:33:49] -- Digit timeout set to 3 [Mar 29

Originally posted by: mailto: Hi all Now I'm making IVR sequance that is customised [mainmanu]. I wish to notify invaid command like a following exten => i,1,playback('your command is ...') exten => i,2,playback(${EXTEN}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I

I'm a newbie to Asterisk, but I'm moderately knowledgeable about phone systems. Right now, I'm most certainly confused. I have a TDM-04B (four FXO) and four analog FXO lines running into it from an AdTran 616. I have Asterisk working internally, although I could use some help getting incoming calls to answer properly and configuring my outbound dialplan. Here's where I'm

On 30 July 2016 at 19:32, D'Arcy J.M. Cain <darcy at vex.net> wrote: > > > Not playing the prompt changes nothing. If someone presses '*' while > listening to your answer message then they are in your mailbox. You > better have a password that they need to enter to continue. I have now tested the 'Unavailable' message by pressing "*" while

Hi, I have configured asterisk now with ENUM lookups which are working really perfect. Now I want to play a small soundfile before dial the number to inform the caller which protocl is used (SIP, IAX2 or ISDN). How can I do this? With Playback it doesn't seems to work: [iax2-sipport-out] ; with leading 3 using IAX-sipport exten => s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out) exten

Hi all Now I'm making IVR sequance that is customised [mainmanu]. I wish to notify invaid command like a following exten => i,1,playback('your command is ...') exten => i,2,playback(${EXTEN}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I know to use with gsm filename.

I have the following in the dialplan: [macro-systemrecording] exten => s,1,Goto(${ARG1},1) exten => dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav) exten => dorecord,n,Wait(1) exten => dorecord,n,Goto(confmenu,1) exten => docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording) exten => docheck,n,Wait(1) exten => docheck,n,Goto(confmenu,1) exten =>

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

Greetings, I'm trying to continue to do some processing after a TIMEOUT (absolute). In my dialplan below, when a call comes in to [default], I call macro-phonenum and pass it a timeout of 20 seconds. macro- phonenum sets TIMEOUT(absolute), then loops saying the phone number that was called (in MACRO_EXTEN). When the timeout expires I want to call my macro-hangup (so it can say

Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have

I've setup my IAX2 over FWD and it is working I can receive a test call and I can call out. Though I cannot figure out how to dial 1-800 numbers over FWD When I dial 1-800 it hangs up on me. Here is a typical session: Called xxxxx:xxxxxxx@iax2.fwdnet.net/18007425877 -- Call accepted by 65.39.205.121 (format ULAW) -- Format for call is ULAW -- Hungup