Displaying 11 results from an estimated 11 matches for "radiancetech".
2004 Jul 14
2
RE: [Asterisk-User] asterisk compile problem
From: "Nik Martin" <nmartin@radiancetech.com>>
To: <asterisk-users@lists.digium.com>>
Subject: RE: [Asterisk-Users] asterisk compile problem
Date: Wed, 14 Jul 2004 09:22:38 -0500
Organization: Radiance Technologies, Inc.
Reply-To: asterisk-users@lists.digium.com
Fletcher Bonds wrote:
>> Hello all
>>...
2004 May 18
1
VoiceMailMain dumps user back into my incoming context after leaving a message
...1. ParkedCall(717)
[res_parking]
'718' => 1. ParkedCall(718)
[res_parking]
'719' => 1. ParkedCall(719)
[res_parking]
'720' => 1. ParkedCall(720)
[res_parking]
Nik Martin
Lead Software Engineer
Radiance Technologies
nmartin@radiancetech.com
W 251.445.0045 x105
C 251.455.4665
F 251.445.0046
2004 Jun 17
1
VOIP wiretapping article
Of course, big brother wants his say in the matter.
http://www.wired.com/news/politics/0,1283,63884,00.html?tw=wn_2polihead
2004 Jun 08
0
Camp On configuration?
Is there a clever way to camp on an extension in asterisk? What I need is a
way to answer my extension (not just a ringing ZAP channel) from any other
phone. If I'm in another office and hear my phone ringing, I want to be
able to quickly pick it up from that extension. The list revealed the
pickupgroup parameter, but that looks like it will pick up any zap channel
that's ringing. This
2004 Jun 10
1
Manager logic to pickup a ringing extension
Can the Manager Redirect command transfer a ringing SIP extension? I'm
trying to implement a Camp On feature, and having failed to do it in Dial
Plan logic, am trying to do it with manager logic. If an arbitrary Sip
extension is ringing, I need the ability to pick up that extension from any
other phone. What little docs there are on Manager commands shows Redirect
takes these parameters:
2004 Jun 14
0
If IAX client is not logged in/registered, Dial plan executes BUSY vs UNAVAILABLE
If I have an IAX client (Firefly softphone in this example), and the client
is not registered at the moment because they are not connected to the
network and someone dial that extension, they get the user's "I'm on the
phone at the moment" message vs. the "I'm unavailable" message. Is this by
design?
Here's the extension in question's dialplan:
2004 May 21
2
Asterisk upgrade on production box
What is the best way to upgrade a production asterisk box? make upgrade? I
don't want my configs messed with, and need the process to go as smooth as
possible. I fetched and built a new kernel last night, but haven't rebooted
into it. I'll do that tonight, and then want to quickly upgrade to the
latest asterisk (mainly for zttest.)
Does make upgrade fetch head?
Thanks
Nik
2004 Jun 04
2
(possibly) new use for asterisk
Has anyone ever thought configuring asterisk on a pair of pc's to act as
remote broadcast terminals for the broadcast radio industry? Seems like
a stripped down asterisk on a laptop with a PCMCIA ISDN modem connecting
to another asterisk instance on a PC at a radio station would work
nicely. Use one of the higher quality codecs, interface the remote
mixer to the sound card on the
2004 Apr 13
4
Dial Plan Format Strings
In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd
like to do something possibly unique with the formatting of extensions in my
dial plan, and am having trouble. We use VoicePulse connect, which gives us
local DID for inbound and outbound calls (even though DTMF tones are not
working in Voice Pulse Connect at the moment). To dial local numbers, you
have to
2004 May 25
6
Downgrading Asterisk
I upgraded to the latest HEAD version of asterisk, and all IAX calls started
sounding choppy. It was suggested on the IRC channel that I go back to
asterisk -stable to determine if that fixes it. Is downgrading as simple as
upgrading? Because now, -stable builds fine, but I get an error on the
asterisk console when starting, something about "ast_get_txt" not found.
Recompiling and
2004 Jun 09
0
Call Pickup problem in Asterisk with SIP phones
I'm having a tough time getting call pickup to work on *. Here's my
configuration:
X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image
A call comes in, and * picks up and presents a menu. Caller chooses
extension, (in this case ext 103, SIP/wsmith)
Wsmith is sitting in my office, hears his phone ringing, picks up my phone,
gets dial tone, and presses