similar to: Caller entered digits ignored during wait....

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

Hi to all, I and using asterisk with following 1. TDM400p card with four FXS modules, So there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 But when some dials form his number (suppose abc) to my number (suppose xxxx) I get abc number on my analog phone, but now I have purchased more than one numbers

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 =>

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

Does anyone know if there are any plans for Zapata or anyone else for that matter to come out with a 3.3v PCI version or PCI-X version of those cards? Gene

When I attempt to start asterisk with my modem setup listed it will not start attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated. running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org) just e-mail me privately if you need more info Thanks in

I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press "#" to end the recording, at which point I am told "Your message has been

Hi, I've looked at the Wiki for this, have seen the Swift.agi details, but has anyone got a current script for Cepstral and an example of integraton in * please? I'm a * and linux newbie, so please be gentle ;-) Thanks John

Hello All, I am trying to set up faxing using Asterisk@home 0.6. I have followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing

Greetings, I am running Asterisk with a T100P and a Zhone channel bank for over a month now. For the most part it works fine but from time to time (about once a week) the system will not let go of a line and will play the greeting over and over. Anyone calling gets a busy signal. If I reset Asterisk everything works fine. Has anyone seen this problem before and fixed it? If so what did you do?

I am trying to configure MeetMe so that external callers can enter the conference rooms after an IVR menu. I have created Conf rooms for all internal Ext's with a prefix of 8. When I call into the system from my vonage trunck the IVR picks up but will not let me dial a conf room. It tells me it is a invalid extension. Can anyone help with a sample conf on this? Thanks, RC

Greetings, This may be a bit arcane but does anyone know what the contents of a facility message should be for initiating a TBCT on an NI2 ISDN. I've been trying to get it to work on a DMS100 for the last four months to no avail. The message I am currently sending makes it to the switch but is returned with unknown message. Perhaps someone here has done it before and can help me out.

Using an example provided by "The Hitchhiker's Guide to Asterisk", I made the following addition to my extensions.conf file: [inbound-analog] exten => s,1,Wait(1) exten => s,2,SetVar(counter=0) exten => s,3,Answer() exten => s,4,Wait(1) exten => s,5,DigitTimeout(15) exten => s,6,ResponseTimeout(10)

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and

Hi All when I try to use auto-dial to connect to outside phone , my applications get executed before the caller attend the calls , this happens only when I call outside no , ie when I use Channel: ZAP/1/0507451111 in my sample.call file , if I use Channel:SIP/326 , it works fine my ?sample.call? file contains Channel: ZAP/1/0507451111 Callerid: Asterisk MaxRetries: 2 RetryTime: 10

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

I'm trying to set up the following application: When a SIP extensions calls another one which is busy, the caller would be able to ask for an automatic callback: when the callee becomes available again, asterisk would ring both the caller's and the callee's phones and connect them when both parties answer. Has anybody done this before? (I tried to search the archs but couldn't

Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say "successfully", I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to

I have this code which was taken from the phpagi project page along with the following in extensions_conf and the output from the asterisk CLI. When I call the 311 extension, I does nothing then hangs up. What am I doing wrong?? ----php code------------ #!/usr/local/bin/php -q <?php set_time_limit(30); require('phpagi.php'); $agi = new AGI(); $agi->answer(); $cid =