similar to: Feedback request: AGI GET DATA change termination digits

I'd like some feedback on potentially submitting a request (and probably a patch too) to change the way the AGI command GET DATA works. Right now, # terminates the entry, which is then returned with the # stripped off the end. What I'd like is to allow user configurable termination digits, which are not stripped off the end. Reasoning: Some entries you'd like to terminate with #.

> -----Original Message----- > From: Steven Critchfield [mailto:critch@basesys.com] > On Wed, 2005-02-09 at 09:15 -0700, Kevin P. Fleming wrote: > > David Brodbeck wrote: > > > > > Anyway, I figured it out. The extension was working, but > Background() > > > ignores the tones A through D by default. I didn't > realize this because I > >

Hey.. Your saying I can not use flash with ISDN ? What options to I have to transfer a call directly ? ( So I have a free line afterwords) >> What interface are you using? ZapBRI? if so you might be able to do the >> hairpinning if it is supported. Im not using any interface.. But if you know how to do that, let me know and I install that interface. Thx for your answer :) Gr?sse

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Forwarded back to the list so others might get the benefit of the answers, and I get fact checked by others. -------- Forwarded Message -------- > From: Lee <leeb00@gmail.com> > Reply-To: Lee <leeb00@gmail.com> > To: Steven Critchfield <critch@basesys.com> > Subject: Re: [Asterisk-Users] udev or not? > Date: Fri, 10 Dec 2004 13:00:29 -0800 > On Fri, 10 Dec 2004

Just change the install prefix directory in asterisk makefile to something like /usr/local/asterisk, and then you will see all the files needed for asterisk to run. > -----Original Message----- > From: Steven Critchfield [mailto:critch at basesys.com] > Sent: Saturday, 1 March 2003 22:03 > To: asterisk-users at lists.digium.com > Subject: Re: [Asterisk-Users] How could I install

Okay, so like this? PHONE1+AD0-SIP/2000 PHONE2+AD0-SIP/3000 PHONE3+AD0-SIP/4000 ALL+AD0AJAB7-PHONE1+AH0AJgAkAHs-PHONE2+AH0AJgAkAHs-PHONE3+AH0- Then you would have Exten +AD0APg- s,1,Dial(+ACQAew-ALL+AH0-,20) Is that right? I have read about the Macros but don't understand their use. Could someone provide an example? Sorry about the newby questions... This will hopefully be my

PoE, or 802.3af, uses a device detection routine to determine if the connected device needs power. The process, in greatly simplified terms, is as follows: 1. Detect link state 2. Send a pulse of a known frequency and intensity over the TX/RX pairs 3. Listen for reflection. 3a. No reflection- provide power 3b. Reflection- no power Devices that comply with 802.3af have filters designed

Speaking as former Dialogic/Bayonne user who was frustrated for months with Dialogic's complexity and months of initial testing with GlobalCall only to use their many-years-old and very complex base Dialogic drivers(and eventually scrapping it all for Digium/Asterisk and being up one week later), I can tell you that Digium/Asterisk is definately the way to go. It is 1000 times easier to

Juan, I dont see people here with "attitude" or any need to shut down this list as you suggested. Contrary to your experience, I find people to be quite helpful here. Steven's comment "Start here and then ask a real question" is definitely not arrogant IMHO. He is just trying to knock some etiquette on Luan. Besides, he did give him all the answer Luan was looking for.

Hello all, I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use "Get Data filename, timeout, maxdigits " in the agi script. In * console I get message saying playing filename but it

Steven Critchfield <critch@basesys.com> wrote: __________ >On Wed, 2004-05-26 at 14:23, tpanton@attglobal.net wrote: >> On which subject, has anyone else >> got time to work with me on a chan_dialogicGC ? It looks do-able but I am ignorant of how asterisk does threading. > >Do you have GPL drivers for the dialogic card? Follow previous comments >about licensing.

Also lookup AGI The WiKi and via google by using this: site:lists.digium.com <some words> W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steven Critchfield Sent: Friday, March 04, 2005 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Placing a call

Here is an example of a couple of macros that help me where I have a SOHO with a home phone line and a work phone line. If I pick up line 2 my work line I would prefer the call I make to go out my office phone line same with if I pick up line 1 my home phone line I would prefer it go out my home line but want it to roll if needed. So with this little macro it is possible for that to happen.

My thoughts on this after reading Steven's very politically worded reply is that IMO your best bet would be to go with RedHat, I am not going to go into details about the if's, when's, why's, and but's.. I am running Asterisk quite well on RedHat 9 and if you like I have created an install guide for setting up an Asterisk box on RedHat 9 which I can send to you if you are

Thanks for the help. I was able to get my application to load with Asterisk, now I just need to get it to work. After reading your comment, I don't know that I fully understand what's going on as far as the channels and extensions. Are you saying that the MWI is tied to the channel? If that is the case, then for my SIP phones, would the channel be (SIP/XXXX)? If it should be, then

Ronald, Grandstream products have a one year warrantee. If you don't have any luck with Pulver, contact us and we can probably get your phones exchanged. Please don't assume that your experience with Grandstream is typical. We sell a lot of these phones and the overwhelming majority of the purchasers are very happy with their units. The quality has improved tremendously over the last

Hi Steven, I wish, I already have 2 spare TE410p and 1 TE405p. But customer wants to use Analogue and they already installed the lines. Yesterday, I ready about the FXO modules being replaced by Digium, this relaxed me a bit. But as you said, I will have to worry about the ring and hangup detection, cleaner lines, impedance matching echo problems specially that I have not heard of anyone using

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

Hi all: A short question: For example, a=[3,4,6,2,3], obviously the 3rd entry of the array has the maxium value, what I want is index of the maxium value: 3. is there a neat expression to get this index? Thank you! Best, Feng [[alternative HTML version deleted]]