similar to: The Start extension

Hi: I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I have an IP trunk to voicepulse and my outgoing calls go over that. I can also receive calls on that voicepulse trunk and want it to an auto attendant. Everything works except on the following: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have

I have posted soo many times in the past but never recieved even a single reply . seem like you people are ignoring me or either way too busy .. never mind this is my last try . How can record a conversation with asterisk ? I tried to use Record() but dint work for me .. here is what i tried . exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer

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

Hello all! We have a PSTN line with four numbers calling into it. There is distinctive ring on these lines. They are are follows: 1. standard ring 2. short ring 3. long ring 4. short ring, long ring, short ring Based on the information I have been able to find, I have created the following entries in my zapata.conf file, to try and weed out some of the timings: dring1=95,0,0

Hi, I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error "Failed to authenticate on INVITE" trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. Has anyone seen this? - Eric

Hi, My client wants incoming callers who do not press a digit to go straight to the operator. Does anyone have an idea of how this could be done? I've looked for some examples, but I'm still not clear on it. Here's the relevant portion of my extensions.conf: ------- ; Wait 15 seconds for an answer (pick up the local phone) exten => s,1,Wait,2 ; Answer the phone exten =>

hi, i installed asterisk and works very well, the only problem is that when i try to call a direct number of a company that has a normal PBX i got this error: to 10.8.210.153:5060 == Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro) -- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack -- Goto (doisdn,00115601992,1) --

Greetings, Below is part of the contents of my extensions.conf file. exten => s,1,Wait,1 ; Wait a second before answering. exten => s,2,Answer exten => s,3,ResponseTimeout,10 ; Set the amount of time the user ; has to make a selection. exten => s,4,DigitTimeout,5

Hi, Greatly appreciate if some one help me with the application read(). asterisk*CLI> show application read asterisk*CLI> -= Info about application 'Read' =- [Synopsis]: Read a variable [Description]: Read(variable[|filename]): Reads a '#' terminated string of digits from the user, optionally playing a given filename first. Returns -1 on hangup or error and 0

Hi, I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux (and a Fritz Card PnP). The ISDN-BRI is in PTP-Mode (Point to Point "german: Anlagenanschluss") which is enabled within I4L with "hisaxctrl fcpcipnp0 7 1". Everything works fine except that I can not see the called number/MSN of incoming calls within Asterisk and because of this I can not route incoming calls

Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks

Sorry that I wrote in german : Ich benutze asterisk mit dem zaphfc Treiber. Jetzt hab ich folgendes Problem, habe 2 ISDN-Telefone angeschlossen. zaphfc im nt-mode. Anrufe von ausserhalb per sip (sipgate.de) kommen an. Wenn ich aber intern zwischen den zwei Telefonen (Ascom Eurit 30) sprechen m?chte geht das nur wie folgt : Erst die Nebenstelle w?hlen und dann den H?rer am Telefon abnehmen.

Hi all. I am new to the list and new to asterisk. I have asterisk installed and running. I am using it as a voicemail server only. What I would like to do is send users to a general mailbox that will be addressed as <companyname>@asterisk and give them the option to wait for the tone and leave a message, or press 9 to dial by name. My questions are: 1. What is the best way to do

Hello all. I'm having a problem debugging an IVR I'm building. I can't see any reason this shouldn't be working. Firstly the asterisk version is: Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a i686 running Linux on 2006-02-17 22:44:48 UTC Basically the problem is this. While the playbacks are happening you can push any one of the options and to happily

I've had my * setup installed with an X100P card for a couple of weeks and it's great fun! I'm even giving a demo to the local Linux group in a couple of days. But I have a snag. I have the X100P on a shared line, and configured to wait for 20 seconds before answering and doing the auto-attendant/voicemail dance. My problem is I can't find an application command to cancel the

exten => auth_dial,1,DigitTimeout,5 exten => auth_dial,2,ResponseTimeout,15 exten => auth_dial,3,Read(dialed,IVR/en_enter_destination,0) exten => auth_dial,4,agi(call_start.php|${dialed}) exten => auth_dial,5,dial(SIP/${dialed}@146.82.15.241) I'm trying to get What they dialed put into the PHP script. How do I get the contents of this variable in the php script?

We are using DISA with local SIP users. The user enters in a 2 digit code then they get a dialtone and the phone dials out. The problem is that the calls waits 10 seconds after the outgoing number is dialed, no matter what I put for the timeout values. Anyone else using DISA that has run into this? exten => _2X,1,Answer exten => _2X,2,DigitTimeout(2) exten =>

I'm testing an ivr implementation (first time) using: exten => 620,1,Wait,1 exten => 620,2,Answer exten => 620,3,DigitTimeout,5 exten => 620,4,ResponseTimeout,10 exten => 620,5,Background(npi-greeting) ; "Thanks for calling press 1 for" exten => 1,1,Goto(npi-directory,s,1) For initial testing, I've arbitrarily mapped this onto ext 620 (will change that later

Hi, Sorry, in my last mail I wrote "wcfxs" instead of what I actually used, "wcfxo." I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and

Hi everyone: The following code dials our prefix, sends a beep, and sends a DTMF "c" tone, then dials the phone number. I need to send the DTMF only if the phone is answered. [voip] exten=>i,1,NoCDR() exten=>i,2,Hangup() exten=>s,1,Wait(2) exten=>s,2,Background(beep||) exten=>s,3,DigitTimeout(6) exten=>s,4,ResponseTimeout(10) exten=>s,5,SendDTMF(c)