similar to: Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

Hello, I need to write a script which prompts the callee to type a number, and then read it back to them as confirmation: ======= extensions.conf [robocall] ;Expect 10-digit number excluding final #, 2 tries, 20s time-out exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20) exten => s,n,GotoIf($[${LEN(${NBR2CALL})} != 10]?end) ;exten => s,n,SayDigits(${NBR2CALL}) exten

; *99: ; 99 bottles of beer on the wall. exten => *99,1,Noop(99 Bottles of beer on the wall) exten => *99,n,Answer() exten => *99,n,Set(bottles=99) exten => *99,n(loop),Noop(There are ${bottles} bottles of beer on the wall) exten => *99,n,SayNumber(${bottles}) exten => *99,n,Noop(Take one done and pass it round and there's) exten =>

Hi all, Is there a "speed dial" type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar

Hi, I created a dummy dialplan where I ask the user to enter the age. [macro-age] exten => s,1,Background(my/age) ;;Play recorded message to enter age exten => s,n,WaitExten(10) exten => _XX,1,Set(AGE=${EXTEN}) ;; this line is not executing, instead dialplan is terminating with error given below. exten => s,n,NoOp(${AGE}) exten => s,n,GotoIf($[${LEN(${AGE})} >

Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any "messages index file" or there isn't any file to index them? Could you share any script to do that? Thanks in advance. VoipCrazy. -------------- next part -------------- An HTML attachment was scrubbed... URL:

This might be a newbie question... I'm using a SIP trunk and trying to get DID line information on an incoming call. All I hear is a nice lady saying 'Zero' - then the call continues... Any suggestions? thanks Todd exten => s,n,Set(DIDID=(<${FROM_DID}>)) exten => s,n,SayNumber(DIDID) or exten => s,n,Set(FROM_DID=${EXTEN}) exten =>

Hello - I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am having problems with routines that input long strings of numbers, in that I am getting more than a small number of double digit entries. As an example, I have a section that asks for the user to enter a call forwarding number, and then puts that number into a database. Almost always, there are double digits when the

Hey all, Working slowly on getting the myriad number of parts to my fax system plan together, and one of the pieces I want to nail is how to go about, for the outbound context (fax-out) pre-pending a digit onto a number? I.e., for all my testing right now, I've been dialing '91XXXXXXXXXX', as the asterisk server doing faxing junctions into my old Rolm CBX switch, and so I need the

hello i want to kick participant in a meeting by pressing the digit on sip phone.when i entry the meeting ,no matter how i press the button,the dtmf does not work. here is my dialplan and my agi script,and sip.conf [from-internal] exten =>121,1,MeetMeCount(900,CONFCOUNT) exten =>121,2,GotoIF($[${CONFCOUNT}<10]?3:100) exten =>121,3,Authenticate(123456) exten

Hi there, AGI is partially not working for me with SAY DIGITS, SAY NUMBER, WAIT FOR DIGITS etc. It appears that result is always one for any function that looks for user input, regardless of which key was pressed. Playing sound is only possible using EXEC. This applies to two * servers with RH 7.2 and RH 7.3 and very recent CVS. Any suggestion or tips about where I goofed? Thanks, Philipp

This is a follow/find me script that I can't quite get to work, asterisk wont save forward/${calleridnum} to AstDB... any comments or thoughts on how to make this work or change it to work differently are appreciated. The voice prompts to go with all playback/background extensions are commented appropriately. I hope this code is of use to some of you and any help with a perfected

Ok, probably a dumb question. I believe I already I know the answer, but thought I would get feedback from others. One of the issues with user devices at the end Asterisk is dialing time out. This is a parameter within each hardware device. So if I set it to 3 seconds it appears from the moment after going off hook any key press starts a timer allowing me 3 seconds to enter the next number

Hi, Currently, it seems impossible to use the output of SayNumber application as an input to Read application. So, if you want to develop an IVR with something like "You've got 23 messages. Type 1 to listen to the first one. Type 2 to leave", you must parse this message into 3 pieces and want for the last one play to start listening of user input : Playback ( "You've

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

http://bugs.digium.com/bug_view_page.php?bug_id=0001429 * Support for other language syntaxes in saynumber Accidentally I opened this can of worms to see if we can add support for other language syntaxes for saying numbers. Seems like Swedish, english and norwegian follow the same syntax. I've integrated existing patches for french, danish and soon portuguese syntax. The steps we're

Hello, I use the SayNumber() with variable. for example the number 1234 - asterisk play the number without and. -- Executing [888 at from-internal:1] Set("SIP/103-0000035d", "LANGUAGE=en") in new stack -- Executing [888 at from-internal:2] SayNumber("SIP/103-0000035d", "1234") in new stack -- <SIP/103-0000035d> Playing

Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan

Hello, with Asterisk 1.6.1.6 I try to hangup a call if called extension is not existing. For this purpose I would use the internal i extension but seems not to work. [MyContext] exten => s,1,NoOp(Call is treated as it should) exten => s,n,NoOp(next step) exten => s,n,NoOp(aso ...) exten => _[a-zA-Z].,1,Goto(s,1) ; accept exten LEN >1 alpha exten => _X.,1,Goto(s,1) ;

How about: exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for the remaining 20 of the 40 seconds, as the Playback will not answer