similar to: How to get ten-digit number?

Hi The first step I have to go through when users call into our IVR is to handle the case where users' PBX hides their CID number. In that case, I need to have them type their phone number (ten digits). OTOH, those who call without hiding their CID number are sent directly to the main menu. How would I go about prompting users for their phone number? Here's what I have at this point:

Hi, why would Asterisk core dump with the following test dialplan extension ? exten => 8100,1,Answer() exten => 8100,n,Set(CALLERID(all)="") exten => 8100,n,PrivacyManager() exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:nocid) exten => 8100,n,NoOp(Number is ${CALLERID(num)}) exten => 8100,n,Hangup() exten => 8100,n(nocid),Playback(vm-goodbye) exten

I wish to have my Grandstream GXP-2000 phones make a different distinctive ring for internal calls ( Internal ) or if the incoming call has no caller id 'NOCID'. The Grandstream phones calls allow 3 distinctive rings depending on the caller id. I have one set up and working for 'Internal' calls but unfortunately the same tone will ring if caller id is absent on a call. My

Hello I didn't find the answer in the ATOF 2nd Ed: When using the Record() application, I need to know how it ended: Did the user leave a message, or did he hang up? If the latter, Asterisk stops right there, while I need to run some other commands before hanging up: ======== exten => _[1-4],n,Playback(/root/asterisk_sound_files/leave_msg) exten =>

Greetings everyone, today I spent the last part of my day trying to find a parse error inside this snip: http://pastebin.ca/740081 If there's anyone who can shed some light on why my GosubIf condition is throwing a parse error, I'd greatly appreciate your insight. This was really frustrating and is probably a stupid mistake. Regards, -Michael

Hello About Record(), ATFT 2nd Edition says that "if the filename contains %d, these characters will be replaced with a number incremented by one each time the file is recorded." Problem is, the documentation doesn't explain how to refer to this filename later in the dialplan :-/ In this particular example, I want to move the file to the web server's /htdocs so users can

I am having trouble understanding how to use this. I want to transfer certain incoming calls from an IAX ITSP based on caller ID. From what I can make of the docs, I thought I need to do something like this... exten => _NXXNXXXXXX,n(nocid),transfer(1000) exten => _NXXNXXXXXX,n,noop(boo,${TRANSFERSTATUS}) exten => _NXXNXXXXXX,n,hangup exten =>

Ok, that's done. Now after this the should compile and run : in En.rc MAIN_MENU is defined now like this : MAIN_MENU MENU { POPUP "&File" { MENUITEM "&New...", 0x100 En.rc is included by #include "En.rc" in rsrc.rc. And gcc complies about it when compiling : [syl@snoop notepad]$ make gcc -c -I. -I. -I../../include -I../../include -g -O2 -Wall

I am using asterisk 1.6 at home and would like to send incoming calls without caller id immediately to voicemail (i don't want to use the privacy manager where people have to enter a number). The config examples i found are all for the pretty obsolete 1.0 and 1.2 versions of asterisk. Would anyone be willing to share a config example? Thanks!

Hi all, My latest Asterisk system is based on Debian squeeze with Asterisk 1.6.2.6-1 and SIP only. One of my favorite features that I had working with Asterisk 1.4 is the PrivacyManager. However, this was not straightforward, because anonymous SIP calls arrive with ${CALLERID(num)} = "anonymous", instead of being blank. So, to get it to work I added the first three rules to

Hi all, Previously, I had the PrivacyManager working for me exactly as would be expected, but after upgrading the OS to Debian lenny and Asterisk to v1.4.21.2 that's no longer the case. Anonymous callers are still confronted with the PrivacyManager, but now no matter what I set the minlength value to, e.g.: exten => jaap,n,PrivacyManager(1,1) ... (I'm not using a

I am having issues with Privacy Manager and Zapateller. If I set callerid="" on a sip user zapateller sends the tones If I set callerid="Anonymous" <8475551212> zapateller doesn't send the tones If I call from a phone after dialing *67 zapateller doesn't send the tones In the last 2 cases, the display on the phone shows -Blocked Call- PrivacyManager always gives

Hello Doug, Friday, December 13, 2019, 11:03:37 AM, you wrote: >> This is exactly what I do - “press 1 for a human” >> Works great > I do this as well, but I also do a database lookup to see if the number > is on our speeddial list and if so, pass the call directly on without > the IVR prompts. I do something similar for calls without caller ID, but I was still getting

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am running into some problems here with PrivacyManager. We used to use it without any issue, but now there seems to be several problems. We are currently running Asterisk 1.2.4. First, it seems that if the user does not press the pound (#) key after entering their number, PrivacyManager will fail. I have the minlength set to 10, and

Hi All, In theory if I do this; exten => s,1,Zapateller(nocallerid) exten => s,2,Privacymanager exten => s,3,Dial(a bunch of SIP extensions) My callers should only hear the anti-telemarketing tones if they call from a line that has no caller*ID and then get offered an opportunity to enter it, right? What I'm finding is that in the event of no CID the caller gets dumped into the

Folks, I'm trying to implement a simple loop in a dialplan. The object is to set a counter, run through some IVR options, increment the counter, return to the start, then finally fall through to an operator or voicemail. Am using 1.4.10 and have reviewed doc/ exten => s,1,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=20) exten => s,n,Set(loop = 0) exten =>

I have been having problems getting PrivacyManager to work correctly. Right now I am running the 1/21/05 CVS but I have been unable to get this to work on asterisk-stable either. You can see from the debug below that the inbound call is arriving via IAX2 and both the CALLING NUMBER and CALLING NAME are both set as "Unavailable". However, PrivacyManager executes and determines that

Hi, I was trying out call file just to see how they worked and my system does not seem to do anything with them, although asterisk *is* deleting the files that I put into /var/spool/asterisk/outgoing. 1. I nano'd a quick call file like so: Channel: SIP/axVoice/9105555555 CallerID : Leebo <5555555555> MaxRetries: 2 RetryTime: 30 WaitTime: 10 Context: main_menu Extension: s Priority:

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS

Hello List, Im just dipping my feet into the asterisk world, and im having major fxo problems Im running Asterisk (from svn) + libpri (from svn) + asterisk-addons (from svn) + asterisk gui (svn 1.4 branch) + zaptel (svn 1.4) on a Debian Etch box, with 1gb ram, running all of the services for my home server (web / db / music server etc), and i would like to run my PSTN line from Kingston Comms,