search for: hazelbaker

...ponds with 404 and nothing is logged. Can * be used in this manner in a dialplan? If so then any suggestions on what I can check to see why it is doing this? If not, does anybody have a better suggestion for me? I'd rather not use a regular digit as the "begin" code. Daniel Hazelbaker

...seen that the polycom setup (601+sidecar) works but only for up to 7 phones. Does anybody have a recommendation for a solution for this? I find it hard to believe that nobody makes a compatible phone (or add-on) that is compatible with Asterisk. It seems like such a common thing. Daniel Hazelbaker

Is there a way to force a channel to continue in the dialplan after the remote end hangs up? Specifically, I am trying to play around with setting up a fax server. I can receive the fax, but sometimes the sending fax hangs up before my System command for printing can run and the fax never prints. I know I can work around by setting up a custom context and use the 'h'

I am trying to get the presence/hints/BLF working along with Realtime SIP but I never get any "busy" notification. core show hints always shows the realtime sip user as idle. I have tried setting call-limit to various values, including 1 but nothing seems to help. I have tried limitonpeers both yes and no. Anybody got any other ideas? I do know the hinting is working as I can

I am suddenly getting a bunch of OLD (as in 3-9 months old) e-mails from mantis saying things like a note has been added to an issue etc., and yet the issue has not been touched in months and the "new note" it is referring to is also months old. Consequently, I never received these e-mails before either. The e-mail itself shows that "carolina.digium.com" received

...t and we don't want to end up with phones that don't work properly with asterisk. 3Com 3101 (model with speakerphone) 3Com 3102 Business Phone 3Com 3103 Manager Phone 3Com 3105 Attendant Console (if these don't work, can somebody recommend another receptionist alternative?) Daniel Hazelbaker

...lar users to modify /usr/local/bin. So people could install other programs that asterisk runs into that directory as well to get elevated privileges. For me it is not a concern as the machine is used only for Asterisk and only accessed by our IT department. Daniel > CP > > Daniel Hazelbaker wrote: >> On Apr 1, 2008, at 5:22 PM, asterisk-users-request at lists.digium.com >> wrote: >> >>> Can the volume of the recorded voice mail message be changed? If >>> so, what I am doing wrong? Any input would be greatly appreciated. >>> Thanks. >&g...

Does anybody know how big a presence Asterisk and/or Digium will make at Networld Interop this year? I have a part-time guy that is building an Asterisk system for us (in a proof of concept fashion before we do a full switch to it) that I would like to take, but I don't want to waste his time if it is going to just be a "yeah, try our product" booth and not something he

We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled

Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0

We have a Asterisk box acting as a voicemail system and greeting/ call director for our phone system (NEC system). The problem we are having is that randomly (though most especially with cell phones) asterisk thinks it is getting a double digit. For example, somebody will enter "269" and asterisk will read "2269". I believe the core problem is the NEC system's

...ine so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Alarm on Unicall (acriollo) > 2. Ability to put call on hold via manager? (Steve Totaro) > 3. Re: Receptionist Phones (was 3Com Phones) (Daniel Hazelbaker) > 4. Call Waiting Issues (Brad Glonka) > 5. Wanted: Cd-bootable Fedora+Asterisk (Bruce Komito) > 6. Master.csv Shell Script (Jeremy) > 7. Re: Ability to put call on hold via manager? (Alberto Sagredo) > 8. TE 205P/A102 fit in hp dc7600? (JOSE MANUEL CORTES DAVID) >...

Hi, For a long time, I was wondering if I should use MAC address instead of Extension number to identify SIP endpoints (as I'm mostly not using softphones). Before diving into this, I wondered how people using MAC address are using CLI as it seems more natural and simple to type "sip show peer 4566" as opposed to "sip show peer 00147F784512". Is there something obvious

HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and

I've been playing with Asterisk for a few weeks and it's working great. I have a question about getting multi-line receptionist phones working. I was thinking about getting one of these expansion ports: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0 9186a008008883d.html What are people using for receptionist phones that show all the extensions in

Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't