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Hello, I think I speak for many people here when I say we'd love to see the specifics of how you have your Asterisk network set up (phones/server hardware/Asterisk setup/...). MATT--- -----Original Message----- From: Jared Smith [mailto:jsmith@drgutah.com] Sent: Thursday, January 08, 2004 11:50 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk success stories in small-mediumoffice environments? On Thu, 2004-01-08 at 09:09, Jeffrey Paul wrote: > I'm not really looking for working configurations as much as I am...

> -----Original Message----- > From: Jared Smith [mailto:jsmith@drgutah.com] > Sent: Monday, 12 January, 2004 10:41 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Turning a profit (WAS: More words > for Allison) > > > On Mon, 2004-01-12 at 04:49, Alastair Maw wrote: > > Hmmm... I think John's turning a profit... :)...

Thanks we just figure it out a bit ago. It's amazing how simple some things are when you just ask - and then realized that you were making it too hard to begin with!! :-) Lane Hoskins, MCP Network Engineer 540.767.7626 -----Original Message----- From: Jared Smith [mailto:jsmith@drgutah.com] Sent: Tuesday, January 13, 2004 10:59 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] inbound call routing problem On Tue, 2004-01-13 at 07:52, Lane Hoskins wrote: > We have 8 lines coming into an ADTRAN channelbank that then goes to > the * server via a T100P card...

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

I have an X100p card....and it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael

I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented

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After reading about these $75 SIP phones on this list, I purchased a couple for evaluation. They do work with asterisk - and are good value for money, but as somebody commented: they are not yet perfect. I just wondered if anybody had managed to get either the message-waiting indicator or the conference button to work? Phil Skuse <phil.skuse@vicorp.com>

Mahmut, excellent summary :-). I look forward to your next update. One little thing, In the manager events that show start/stop monitoring, can you please include a field that indicates the filename(s) to which the monitoring was written? Thanks, Ben -----Original Message----- From: Fettahlioglu, Mahmut [mailto:Mahmut.Fettahlioglu@oa.com.au] Sent: Tuesday, April 15, 2003 5:17 AM To:

First of all it works great!! I have my 3Com cards booting off the network and it grabs the linux kernel just fine. My next problem is more of me not understanding the rules. Is the /tftpboot directory just like /? My though was to take a Linux rescue disk that had some imaging tools and then just put the whole thing in /tftpboot and tell it to boot the kernel in that directory. But I always

Does anybody have or know of a mini linux distro that I can boot from PXE? Allot of this is fairly new to me so I'd rather not re-create work that's already been done. Basically I'm just trying to get to a command line, and have the ability to get to FTP, SMB, or NFS. I've been messing with pxelinux.0 and I can load a kernel and all that just fine, but I can't seem to get the

Has anyone encountered this SIP problem? I dial a SIP phone from a ZAP phone and after talking for a few minutes the SIP user transfers the ZAP user to another ZAP user. The transfer works correctly, but when the native bridge occurs -- Attempting native bridge of Zap/22-1 and Zap/16-1 the music on hold never stops. The two lines are bridged correctly and you can hear one another but you

I upgraded to the latest and greatest from CVS today, and now SIP pickup groups appear to be broken. Can anyone else tell me whether or not they're seeing the same problem. If anyone out there can verify the problem, I'll submit it as a bug. Jared Smith

I know this is going to sound like a strange question, but here goes: Does anyone know of a SIP softphone that has either a button or a programmable soft-key to send the current call to VoiceMail? I'd appreciate any ideas you might have. Jared Smith