On Thu, 5 Feb 2004, John Todd wrote:
So, to boil your problem down to what I think is the problem:
When you attach an inbound call to the DISA application, it does not
produce a dialtone fast enough.
snip
[main1]
;
; Take any number, and give it to the DISA. The DISA
; just then takes
Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly
straightforward. The release notes indicate that you can trigger other
ringtones on the phone (in the section Support for SIP Alert-Info
Header), but I can't get anywhere with it.
...the Alert-Info header consists of a name of an
(${DelayedChannel})
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On Sat, 14 Feb 2004, Philipp von Klitzing wrote:
Hi!
We need to implement the following:
Call comes in, ring ZAP/1 (6 rings)
For the last two rings, also ring ZAP/2
[incoming]
exten = s,1,DIAL(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED],18)
[test1]
exten =
On Sun, 15 Feb 2004, Chris Clifton wrote:
Can anyone offer adivce for connecting * to a merlin legend ?
I'd like to use a t1 interface to connect the two, * will be used as a long
distance voip gateway in this scenario. Is this possible using a digium
t100p ?
Is it possible? Absolutely.
You'll
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On Thu, 19 Feb 2004, Chris Hirsch wrote:
Bisker, Scott (7805) wrote:
Buy SmartNet support for the phone. That grants you access to software
images through their website. Try Insight. 1-800-INSIGHT. They sell
all quantities.
Given than I'm interested in getting a Cisco phone off something
to a phone (or
an X101P)
The E-10 and E-20A type devices connect to a CO line (provided by telco or
TDM400P).
It's a question of the application - the latter of the two is just a
speakerphone (what you're looking for).
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On Thu, 26 Feb 2004, Greg Kedrovsky wrote:
In a nutshell: Can I use Asterisk to hook up an intercom at my front
gate? My wife would like to have one of those simple
speaker/microphone intercoms. People show up at our front gate, press
the doorbell, it rings in the house. We pick up a phone on my
On Thu, 26 Feb 2004, Rob Fugina wrote:
On Thu, Feb 26, 2004 at 10:19:08AM -0600, Greg Kedrovsky wrote:
On Thu, Feb 26, 2004 at 10:28:37AM -0500, Steve Creel wrote:
As for hardware, take a look at:
http://www.vikingelectronics.com/products/doorentry/product_list.html
Nice. Thanks. I
On Tue, 2 Mar 2004, Emanuele Laface wrote:
My actual pbx is connected to the external world with 2 PRI interface, my
idea is to insert asterisk in the middle, I want disconnect the two PRI,
connect them to the asterisk and connect the asterisk with old pbx with a
cross cable.
So, at the first
(though haven't tried it), you can use:
dacs = 1,3-5:25
to take channels 1,3,4,5 and put them on 25,26,27,28
One note: you can only use dacs on T1/E1 spans, not the pci fxs/fxo cards.
Hope that helps...
Steve
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On Tue, 2 Mar 2004, Emanuele Laface wrote:
On Tue, 2 Mar 2004, Steve Creel wrote:
Your other option is to terminate the two PRIs into asterisk, and use
asterisk to provide two PRIs into your PBX. This gives you access to the
actual call routing.
Ok, my problem is exactly how I can do
On Tue, 9 Mar 2004, Jonathan Moore wrote:
I have seen conflicting references in regard to this. Seems like the Cisco site
has comparison charts that show this phone doesn't have a SIP image, but after
seeing the post I did a little searching and people seem to have a way of
running SIP on them.
On Wed, 10 Mar 2004, John Fraizer wrote:
For what it's worth, I don't have any delay between answer and audio with my
asterisk server and 7960G either originating or answering. It doesn't
matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's
pretty much instant (not
On Wed, 17 Mar 2004, Tim Sailer wrote:
On Wed, Mar 17, 2004 at 08:54:57PM -0600, Matthew Marlowe wrote:
727 or 772? There is 772 in FL available.
727. That's St. Pete/Clearwater. What area is 772?
Tim
http://www.nanpa.com/area_code_maps/display.html?fl shows the Florida area
codes...
For the
On Thu, 18 Mar 2004, Chris Hobbs wrote:
I'm investigating asterisk to use as a replacement for an aging Lucent
PBX in our district office, as well as replacing the Centrex/intercom
based systems at our schools.
I'm curious if any other schools/districts are using asterisk? If so,
I'd certainly
On Fri, 19 Mar 2004, Jacques Leisy wrote:
Sorry for a very stupid question, but I cannot find a supplier anywhere.
Where can I buy the 3 Amps GMT fuses for the Adtran's PSU.
Car fuse don't seems to fit. What is GTM the abbreviation of
A good question (that I wish had been in the archives when
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the Cisco phones. So my question: where do you buy your phones?
We can't buy direct from Cisco (must have 3 quotes).
Thanks...
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think I have to look at mgcp, but was curious as to what others are
doing.
Many thanks,
Steve
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, while having the extension number next to each presentation
(line1, line2, etc)?
Use phone_label for that text...
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will), I'm sure there is someone
far more qualified who could probably write it much better and far more
quickly.
Just my $0.02
Steve
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On Sat, 13 Sep 2003, John Brown wrote:
On Sat, Sep 13, 2003 at 05:21:40PM -0400, Steve Creel wrote:
The Legend currently has 8 analog lines in a hunt group to the
voicemail/auto-attendant system. I went in after-hours and put a buttset
inline to monitor the first line in the hunt group. When
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and then and
adapter.
How are people connecting to large amounts of extensions?
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these phones can have a SIP image (which i downloaded) but
before i upload the image i want to know if anybody tested them ?
Michael
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It's documented somewhere for extensions.conf, and I was delighted to see
that it is a function of the config parser, so yes - it's available in the
other files.
On Tue, 21 Oct 2003, WipeOut wrote:
Steve Creel wrote:
You'll want to #include it. This leaves the burden of the [general] and
any
It was originally announced by Mark in
http://lists.digium.com/pipermail/asterisk-users/2002-March/001766.html
It happens as a function of the config file parser (config.c) and will
work across all of the config files parsed by it.
On Tue, 21 Oct 2003, Olle E. Johansson wrote:
Steve Creel
?
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I have a bunch of existing ATT/Lucent/Avaya 2500YMGP sets with LED
message waiting lights. Do I stand any chance of getting the Adtrans to
light these?
What I know so far:
When you pick up the telephone, the LED flashes.
If you plug two telephones in, picking up one flashes the LED on the
ports. I'd
also suggest upgrading to the latest 750 firmware (L36) as it fixes some
specific MWI issues.
-wade
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Creel
Sent: Friday, March 26, 2004 1:33 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
that supports FSK MWI?
Steve
On Fri, 26 Mar 2004, Steve Creel wrote:
I have L36, and Onhook Messaging is enabled. Does anyone have a reference
for MWI (other than that stuff that turns up on google)?
Make sure that Onhook Messaging is enabled on the Adtran FXS ports. I'd
also suggest upgrading
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I have two Adtran 750's connecting our analog phones to asterisk. On
occasion, I get a channel that gets stuck off hook. 'show channels'
shows:
Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None)
And will just stay like that until the phone is manually picked up and
hung up again (or
On Wed, 12 May 2004, Dan Fernandez wrote:
Folks,
For the last few days I've been trying to experiment with a Panasonic PBX
and an X100P but have run into quite a few problems which I am not sure
if they can be solved with this type of card (how about TDM01B?)
1) I wanted to use *'s IVR
I have a channelized T1 coming in from our telco, terminated onto a TE405.
There are three channelbanks serving internal analog extensions, and about
10 Cisco 7960s.
I have no reports of echo on the analog extensions (as expected). The
7960 users complain of occasional echo (seems like 1 in 5
On Tue, 25 May 2004, Bartosz Jozwiak wrote:
Hello,
I have just received Adtran TSU 600 with 24 FXS ports.
I have installed sucessfuly T100P card.
Sucessfully?
Did you load the module for the card?
What does 'ztcfg -v' show?
Is asterisk running? Does asterisk see the ports? (zap show channels)
Cisco has version 7.1 of their SIP firmware for the 79x0 phones. They
advertise no new software features, but it does include bugfixes for a
number of things. I know there was a discussion about the 0.4sec delay,
which is said to be resolved in this firmware (CSCed48311: Media takes 0.4
sec to
-users/2002-October/005414.html
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luck,
Steve
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On both a 7960 and a 7960G, the 'External Directory' will display only up
to 32 entries. When pressing the 'More' key, there is a 'Next' option.
When you pick 'Next', it does contact the web server, but displays no new
information. The complete directory has 99 entries.
I took a subset from the
to any telephony events, it must be stopped and started (though
'stop now' from the CLI does do the job).
Any suggestions as to where I might look to find the problem?
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We are getting reports from the receptionist that callers are having dtmf
problems (131 gets read as 113, 117 read as 111) - first digit recognized
twice, third digit not read at all).
I have tried 6 different analog phones, two cell phones (on two different
networks), and a digital phone behind
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