Hello,
I hope this isn't too off topic, but I'm attempting to set up a Spectralink
8002 Wifi phone with our Asterisk installation, and seem to be running into a
brick well (more of a wall than others that have posted their experiences). My
problem is that the phone boots, associates with the
. It
then just downloads the first block of each file to compare with what
it already has. If it is the same, it breaks the connection, which
the TFTP server sees as an error. Beyond that, I'm afraid I can't be
much help.
On Fri, Jan 14, 2011 at 2:46 PM, Jonathan C. Bailey
jbai...@co.marshall.ia.us wrote
successfully connected to the TFTP server. What sort of AP are you
connecting to? Could it have a security feature that disallows
reconnects within a certain time frame?
On Fri, Jan 14, 2011 at 3:39 PM, Jonathan C. Bailey
jbai...@co.marshall.ia.us wrote:
I figured that (since the firmware
with Wireshark or something similar?
On Fri, Jan 14, 2011 at 4:15 PM, Jonathan C. Bailey
jbai...@co.marshall.ia.us wrote:
It's a Netgear WG102 AP with a WFS709TP wireless controller. The controller
is basically a rebranded Aruba MC-800 (don't know about the APs).
I've also tried on my WRT-54G
for the info ;)
Sander
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Jonathan C.
Bailey
Verzonden: zaterdag 18 december 2010 18:19
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re
There is a product from Citel (the TVA) that we're currently using with Toshiba
phones. I know they also support Avaya, Nortel, and Panasonic, but am not sure
if they do any other brands. They more or less convert your old digital phones
to SIP.
They have have full compatibility information on
, December 13, 2010 2:47:44 PM
Subject: Re: [asterisk-users] Voice mail distribution - missing messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C.
Bailey
Sent: Monday, December 13, 2010 2:42 PM
Hello,
I seem to be having an issue with voice mail on Asterisk 1.6.2.15 (file
storage). Whenever someone leaves a message that is distributed to another box
(like VoiceMail(100010011002,u)), but the VM never gets distributed to the
intended recipients. Instead, I get the following in the
:18 PM
Subject: Re: [asterisk-users] Voice mail distribution - missing messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C.
Bailey
Sent: Monday, December 13, 2010 12:00 PM
To: Asterisk Users
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C.
Bailey
Sent: Monday, December 13, 2010 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice mail distribution - missing messages
I assume you mean passing the context with the box
-users@lists.digium.com
Sent: Monday, December 13, 2010 2:35:58 PM
Subject: Re: [asterisk-users] Voice mail distribution - missing messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C.
Bailey
Sent
Just saw this after we did the same upgrade... Take a look at the bug below (we
first saw it in 1.8) - it has a work around that you can use...
https://bugs.digium.com/view.php?id=18185
-Jon
- Original Message -
From: sean darcy seandar...@gmail.com
To: asterisk-users@lists.digium.com
No dice on finding a fix for this. I've been looking through the bug tracker
and through the config files and haven't found anything...
- Original Message -
From: Jonathan C. Bailey jbai...@co.marshall.ia.us
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hello,
We recently upgraded to Asterisk 1.8/DAHDI 2.4/WANPipe 3.5.16. This system is
connected to a PRI where the provider requires long distance codes. Normally,
you dial, see progress and hear a tone (call is still unanswered at this
point), enter your code, and it starts ringing as a normal
...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, November 21, 2010 9:33:21 PM
Subject: Re: [asterisk-users] Early audio (long distance codes) not working
after upgrading to 1.8?
On 10-11-21 09:41 PM, Jonathan C. Bailey wrote:
Does
All the Aastra equipment I have so far all has a 00:08:5d prefix.
-Jon
- Original Message -
From: Frank Church voi...@googlemail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 12, 2010 10:41:16 AM GMT -06:00 US/Canada
I know this may be a bit off topic...
I'm trying to play a pre-recorded message to a group of Aastra phones using
multicast paging. I can page phone to phone without issue, but sending from one
of my servers to the phones results in garbled audio. Anyone else been able to
make this work
asterisk-users@lists.digium.com
Sent: Wednesday, March 31, 2010 2:42:14 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-users] Multicast Paging
Jonathan C. Bailey wrote:
I know this may be a bit off topic...
I'm trying to play a pre-recorded message to a group of Aastra phones using
I'm attempting to link calls together in my CDR and would like to try to do it
via the userfield. Is there any way to copy the userfield between calls when
doing an attended transfer? I can't seem to find anything about it searching
Google.
-Jon
___
I've lurked for a while, but I think this is one of my first pleas for help.
I'm having issues where a parked call using the macro below is getting stuck.
Users park the call via a blfxfer key on an Aastra phone. If the call is a
blind transfer, it tries to park the call. If it isn't a blind
=DOCUMENTATION
level 1: uniqueid=1235508550.71744
Jonathan Bailey
Marshall County, Iowa
1 E Main St, Marshalltown, IA 50158
- Original Message -
From: Jonathan C. Bailey jbai...@co.marshall.ia.us
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 25, 2009 8:39:42 AM GMT -06:00 US/Canada
We're using D-Link DES-3028P switches (24 10/100 + 4 gbit). They also have the
DES-3052P which is a 48 port version of the switch. We're paying ~$500, I think
for the 24 port version from Graybar.
-Jon
- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users
Everyone-
We're looking at using some Citel gateways to serve one of our sites (40
extensions, Toshiba phones). I've found that people seem to like the product
from demos, but I was wondering how many have some of the gateways in
production and if they seem to do the job for the long run.
I can't seem to find anything via Google, and haven't seen this before.. What
does a channel listed like Zap/0:27-1 mean? I can't figure out what the colon
signifies. I seem to see channel numbers like these just before the T1 card in
my Comdial switch craps itself.
-Jon
: Tuesday, April 29, 2008 3:21:30 AM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-users] OT: Polycom 3.0
How do they get away with that?
On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey
[EMAIL PROTECTED] wrote:
Try the RPM from Trixbox. If you need something to open the file on Windows
Try the RPM from Trixbox. If you need something to open the file on Windows,
7zip works fine..
http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html
-Jon
- Original Message -
From: Darrick Hartman (lists) [EMAIL PROTECTED]
To: Asterisk Users Mailing
We've been using D-Link DES-3028P and DES-3052P switches. They can supply full
power to EACH port unlike the Linksys switches we've tried. They're also rock
solid from our experience.
-Jon
- Original Message -
From: Hilary Miller [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
My guess is that you don't have any spans set up, or Asterisk doesn't have
zaptel support... Is chan_zap.so loaded?
-Jon
- Original Message -
From: Jerry Geis [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, April 13, 2008 1:27:56 PM GMT -06:00 US/Canada Central
We use Nagios for network monitoring. We've got a check_pri script that should
be fairly universal. It will return critical for any alarm. Feel free to use
the script as you see fit. YMMV - may skin cats, etc (you know the disclaimer
drill)...
#! /usr/bin/python
# Checks PRI status -
We used it in our installation and had some issues. We were passing fax and
modem calls through via the second port as a TDM bridged call. For some reason,
the timing was off even though we explicitly set the timing in the redfone.conf
file. We replaced it with a Sangoma A102d and haven't been
That's surprising.. When I looked at pricing, the Snom 370 was about $50 more
expensive than a 57i for us (the 57i was $205). Also, configuration wasn't too
bad on the Aastra, but that may just be me.
BTW, it also looks like the Snom has support for an electronic headset lifter
on some GN
I'm having (I think) timing issues in relation to bridged T1-T1 calls via
dynamic spans. Fax calls are intermittently working, but voice is fine. My box
has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs
that are hooked to the box come in via a foneBRIDGE2 (dynamic
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