search for: amatechtel

...doubt there would be problems on other channelbanks with older firmwares. Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the first try with current CVS. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: Darren Nickerson [mailto:darren.nickerson@ifax.com] Sent: Monday, December 29, 2003 10:09 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update. As I mentioned in my first post (below)...

Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 Would anyone have an idea on how I would be able to take the mic in on the computer and put it as the "talking party" for a conference room. I would then be able to set up a "listen only" profile for others to get in on. Reason for doing this is for '...

Greetings, I'm trying to setup an option in my greetingmenu that would allow the caller to select this particular option for emergency calls. That option would dial out on an available PSTN line to a cell phone number. Currently it is setup as such exten => 9,1,Dial(Zap/g1/<CELLPHONENUMBER> where <CELLPHONENUMBER> is the number it is calling out to. When option 9 is

...have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: Todd Lieberman [mailto:todd@tlsolutions.net] Sent: Monday, November 24, 2003 9:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Pressing 0 in Voicemail causes * to hangup The Problem: When a call gets into voicemail from Q...

Just cross pairs 1-2 and 4-5 1 <--> 4 2 <--> 5 Never done an E1, but I think it's the same for a T1 4 wire cable. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: bam [mailto:bam@cqm.co.uk] Sent: Monday, December 29, 2003 9:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] E100P pinouts anyone? I'm trying to hookup an E100P to an E1 PRI that I know is working, it drives an Ascend...

Hello all, Like many of you I'm sure have done, I'd like to build an * box at home for testing/educational purposes (and to eventually have one killer fancy schmancy phone system for my SOHO). If I have two standard phone lines in my house, what is the cheapest hardware solution to interface them with *? Thanks! Kris -------------- next part -------------- An HTML

Hi i was trying to call 17009978275 which is my Fwd line on my notebook from Asterisk and i keep getting this message on the console. -- Executing Dial("Zap/2-1", "IAX2/@iaxtel.com/17009978275@iaxtel") in new stack -- Called @iaxtel.com/17009978275@iaxtel WARNING[1150495040]: File chan_iax2.c, Line 4547 (socket_read): I don't know how to authenticate rob to

I've searched through the lists, but can't seem to find a reference to someone that said they had a time and temperature application for *. Voip-info wiki would be a good place for it as a Festival example!!! Can someone remind me where it's at? TIA. Tim Thompson

...T100P card, but it will plug into a channelbank instead of Qwest. Also, depending on how many lines they are going to use coming in, you will be able to use 24-(incoming lines) for ADSI phones or POTS phones for fax machines, regular phones, etc. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: Howard White [mailto:howwhi@vcch.com] Sent: Tuesday, December 02, 2003 12:55 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Configuring new system for a non-profitorganization <bottom response = on> On Tue, 2003-12...

Hello all I just checked out the latest zaptel/zapata/libpri/asterisk/asterisk-addons from the cvs and ran through the entire make procedures. Everything seemed to go fine however now when I attempt to start asterisk, it says ok but it seems to be immediately crashing. The following messages are displayed in my /var/log/asterisk/messages file for the time right around the crash: Dec 29

I have an installation which is currenly using 14 DID Trunk Lines. I need to be able to use Caller ID information and currently it is not available on these lines. Is there another way to access this information? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/62559e22/attachment.htm

Assuming the price of an ADSI screen phone (say, Aastra 390) was the same as an IP screen phone (say, Cisco 7960) and someone was setting up an * server for their 20 employees (each of whom would have either an ADSI or IP phone on their desk), would there be advantages to using the ADSI phones over the IP phones, or vice-versa? For discussion, let's assume that the hardware needed to

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

This newbie has been trying out Asterisk. It has been both a) surprisingly painful and b) impressive in terms of helpful support from other users. Having got two phones to communicate and then got voicemail MWI going (neither painlessly) I decided the next step was to implement call transfer as per nearly all commercial PBX systems i.e. hold call consult another extension either exit and let

Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message -----...

Hi all, Is there any alternative hardware components for multi port FXO cards, other than Single and E1 or T1 level? For example 4 or 8 port FXO card is ideal. Also the price matters. Thanks!

I don't know if anyone had used these boxes yet. I've installed some, but not connecting to an * system. The guys who ordered them said they ran about $600. They are pretty cool in that you designate what the port will be either Data, FXS, or FXO(not fully implemented yet) This is a "NEW" price too, not a refurbed price. I don't know about anyone else, but I've not

> -----Original Message----- > From: john lawler [mailto:maillist@tgice.com] > Sent: Wednesday, January 07, 2004 1:38 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] yet another question on DID trunks > > Hey Steven, > > Sorry to bother you yet again w/ a question on my seemingly endless > quest to get DID trunks setup for a customer. > >