Displaying 20 results from an estimated 4000 matches similar to: "Re: tdm400p static - out of ideas (Jorge Mendoza)"
2004 Jul 07
4
tdm400p static - out of ideas
Hello,
Over the past several weeks, we have been having an intermittant problem with
our deployment of a TDM400P card (4 fxo). ?We have tried many things, and the
problem still re-occurs.
The Problem:
Occasionally (every 48 hours), the TDM400p card will stop answering incoming
calls on ALL fxo ports. ?Attempts to send outbound calls on any Zap channel
will result in hearing a loud
2004 Sep 23
1
TDM400P FXO and Primus TalkBroadBand
Hi all,
A while back, there was a short thread on using the FXS interface from
a Primus TalkBroadBand device (a DLink ATA) as a incoming line for the
FXO interface on the TDM400P:
Primus <--> DLink ATA FXS <--> TDM400P FXO <--> Asterisk
In that thread, a couple of people suggested that this does not work
reliabley, and the ATA FXS <--> TDM FXO link 'goes
2004 Dec 01
3
zaptel and low ring voltage
Hi all,
Several months ago we built an * box with a quad-FXO tdm400p (REV e/f).
>From the get-go, there has been a problem where occasionally (2-3 times
a week) zaptel/* will not detect the ringing on a line. (The call will
ring through to telco voicemail).
The problem is not specific to a single line or FXO port on the tdm400p.
I have 2 theories:
#1 - the ring voltage for some calls is
2006 Apr 17
6
DO NOT REPLY [Bug 3692] New: regression: symlinks are created as hardlinks with --link-dest
https://bugzilla.samba.org/show_bug.cgi?id=3692
Summary: regression: symlinks are created as hardlinks with --
link-dest
Product: rsync
Version: 2.6.7
Platform: x86
URL: http://rsync.samba.org
OS/Version: FreeBSD
Status: NEW
Severity: major
Priority: P3
Component: core
2004 Sep 23
1
send Flash via FXO
Hi all,
We have an analog line from telco, on which 3-way calling is subscribed
to. This line is plugged into an FXO module on a tdm400p.
If an incoming call comes in on this line, can */zaptel send Flash to
telco via the FXO module? If it could, then an incoming call could be
'transfered' to a cell-phone, for example, with a single analog line.
(where 'transfer' is really
2004 Jul 06
2
Uniden consult transfer
Hi all,
I curious to know if other UIP200 users have this same issue:
You flash (XFER button) to consult-transfer a caller to another extension. If
the transfer target party is unavailable (ie: voicemail), there appears to be
no way to get the original caller back.
If it's a known limitation, has anyone come up with a functional work around?
Thank
--
..................................
2004 Aug 13
2
Static on outgoing calls using either X100P or TDM400P
Hello. I've seen several posts talking about line quality using Digium
cards that are sharing IRQs or on machines where X is running but after
trying all of those fixes I am still having a problem with line static
on outoing calls. BTW, calls that are from one extension to another
extension have no static, however, they have occasional clicks and
pops. At any rate, I was wondering if
2004 Jul 28
1
Zap hanging up others zap.
I have Asterisk CVS-HEAD-05/25/04-17:13:22, Copyright (C) 1999-2004 Digium.
Usign exclusively digium hardware.
3 TDM400P cards.
1 4xFXO
1 4xFXS
1 1xFX0 & 3xFXS
When * is attending FXO calls, bridged to FXS calls, natively ofcourse,
at a random time, the call hangus up.
Also, for example, if a call is done, and an other extension hangup,
there are some probability that the other extension
2005 Sep 09
2
AMP 1.10.009 released!
Hello all,
Asterisk Management Portal 1.10.009 has now been released. This
exciting new version has several notable additions (listed below).
The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find
links to the download, install guide, and documentation wiki.
As usual, please use amportal-users mailing list for discussions about
AMP:
2005 Mar 28
4
AMP-1.10.007 Released!
Hello all,
The "Secret Agent" final release of the Asterisk Management Portal is
now available for download:
http://amp.coalescentsystems.ca/
This exciting new release adds a great deal of functionality and
flexibility. Thank you for all the contributions and feedback!
1.10.007
- Added AMP Users (multi-department, basic multi-tenant)
- Added incremental upgrade script
2004 Jun 30
2
Remote SIP client HACK JOB
I couldn't be happier with the simplicity of this - but it's a hack!
Hi all,
I'm currently using a SIP client (BT101) to connect via DSL to a remote
instance of Asterisk.
- Asterisk has a private IP behind my OFFICE router.
- The SIP client has a private IP behind my HOME router.
I'm doing this _without_ the use of STUN or proxy servers.
Here's how it works:
-
2004 Jun 16
4
UIP200
Hi,
We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54).
We've been having some serious problems:
1) All the phones randomly reboot themselves. Typically when trying to
answer or initiate a call.
2) All the phones will disconnect from a calls with the PSTN after 2-3
minutes.
3) The phones are unable to interact with a remote IVR (digit presses
are not received at
2004 Aug 18
1
Three tdm400p's (loaded with FXOs)
Hi all,
Theoretically, I know it's possible, but is any using multiple tdm400ps
(fxo) in single * box? In a production environment? Any gotchas aside
form irq sharing?
Thanks
Ryan
2005 Jul 31
0
[Bug 2933] New: regression with hardlinked devices
https://bugzilla.samba.org/show_bug.cgi?id=2933
Summary: regression with hardlinked devices
Product: rsync
Version: 2.6.5
Platform: All
OS/Version: FreeBSD
Status: NEW
Severity: major
Priority: P3
Component: core
AssignedTo: wayned@samba.org
ReportedBy:
2004 May 31
0
digium card fax detect AND spandsp
Hi all,
I've run into 2 separate problems relating to fax:
1) Using tdm400p + fxo, Asterisk is unable to detect the fax from some
fax machines (from others it can). Using zap barge, I can confirm that
these troublesome calling fax machines _do_ send the fax tone loud and
clear. Are there certain circumstances where I should expect a Digium
card to fail in detecting a fax?
2) Using
2004 Jun 30
2
Ring tone changes when asterisk answers the call
I have a TDM400P card all installed, and I can call out from a sip phone, but
when I call from the PSTN into the asterisk machine, as soon as the Answer()
gets called, the dial tone changes and is sounds like there is a lot of
static on the line.
Below is the part of the dial plan for answering the call.
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,Dial(Sip/pfriedel,20,tT)
2010 Dec 15
3
my scala markdown implementation
Hi,
I have written my own implementation of markdown in Scala. I only later
realized there is already one
( <http://tristanhunt.com/projects/knockoff/> ), but I put quite some
work into mine and I think it is never bad to have alternatives, so I
wanted to release it anyway. I want to use the same BSD License as the
original markdown, but before I put it out into the wild I wanted to ask
2004 Sep 29
1
mid-call echo
Hello all,
When it comes to echo cancellation, we have had great success using
zapata.conf's echo cancellation. This eliminates the echo at the very
beginning of phone calls.
However, we now have a small office scenario where the users are hearing
the introduction of echo several minutes into a phone conversation
(where the first few minutes of the conversation sound fine):
PSTN
2004 Jun 24
0
false hangups
Hello,
We are using a TDM400p with 4 FXOs and SIP phones in a high call-volume
environment. At least twice a day there are complaints of 'dropped calls'.
Examining the debug logs, I see that in each case, an "on hook" event is
detected, followed by the zap channel being hung-up and * saying "BYE" to the
sip phone:
Jun 23 14:17:22 DEBUG[2441232]: Exception on
2006 Jun 26
2
x100p buying advice
I'm looking to get an x100p off ebay and am not particularly familar
with the "life cycle" of the card.. An "Authentic X100P" listing has a
buy it now of $29.95 and says
There are 3 types of cards Asterisk would recognize:
*Screenshots from the official, original driver install
Cheap "OEM X100P","Clones", "Compatibles", Knock-Offs