similar to: Dialogic channel pricing

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.

I found an old Dialogic card in an abandoned PC, that I think is a Dialogic D/41EPCI based on some googling.. The lspci output says: 00:09.0 Bridge: PLX Technology, Inc. PCI <-> IOBus Bridge (rev 01) Subsystem: Dialogic Corp: Unknown device 0529 I'm just getting started with Asterisk to build a SOHO system. Any suggestions for what I can do with this card? It would be

Hi All, I've had a great success these last couple of days on bringing up Asterisk with a load of Cisco 7940 phones and a couple of soft phones as well. I'd really like to try and link this in to my companies PBX so I can call in from the PSTN as well. I understand there is a charge for the module that would allow me to use my old Dialogic board, can anyway provide some more information

Please rest assure that I have been following the * development for a while and understand the value the Digium hardware gives me vs any other vendor. Most of the people on this list probably know whats good for everyone else, but I like to find out for myself (I am not a CNN junky). Now the * site mentions Dialogic as supported hardware at: http://www.asterisk.org/index.php?menu=hardware It

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

Hello, I am very new to Asterisk, so please forgive me for my basic question. I couldn't find a good answer to this on the info boards. I'd like to know if the following is possible and reasonable to accomplish: I'd like to configure a voice recording system using Asterisk and a Tormenta2 Quad T1 card. A co-worker was able to create this system a while back with Bayonne and a

All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards,

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

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

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

Hi, this is my 1.st response to this list, i hope this will work. I tend to agree with Steven since just allowing other termination digits probaly wont solve your upcoming the issues anyway. I use a wrapper around the 'get digit' which allows me to specify that the * digit repeats the menu but maxium 3 times and if the * star digit is used twice in sequence (without other digits inbetween

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

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The

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.

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