Displaying 20 results from an estimated 20000 matches similar to: "Unknown digits"
2006 Nov 30
0
Digium TE405P dtmf issue
Hi Group,
I have an asterisk running as media gateway with a Digium TE405P 2nd Gen rev 2 with echo cancellation. It is interconnected to a telco carrier via ISDN Pri. The voice quality is clear except that sometimes a hear a beep sound that occure around 5 to 10 secs in the middle of the conversation. When I check the logs in the asterisk, I found this.
Nov 30 00:48:38 DEBUG[27705]
2006 Nov 08
1
Delay between DTMF Down & Detected Digit
Good Morning,
I've recently gotten Asterisk installed and configured our IVR using
FreePBX. Things seem to be going well except a few of our inbound
callers are ending up in the wrong place when trying to connect to a
specific extension. The example I had this morning was someone trying to
call extension 212 and getting connected to the Sales queue which is
option 2 on the IVR. I looked in
2004 May 28
0
Problem with digits blending on inbound pulsed digits?
I have a situation where I am receiving DID calls using Immediate Start
Pulse signalling on a Loop Start trunk. The line terminates on a Newbridge
Mainstreet 3624 channel bank, which provides battery etc. The channel is
converted and routed to Asterisk. The lines are configured as follows:
/etc/asterisk/zapata.conf
; Channels 1-24 service MainStreet 3624 channel bank
context=infrom-did
group=1
2005 Jun 28
0
BRIstuff/OctoBRI problem: Ring requested on unconfigured channel 255/255 span 5
Hi all,
I just posted this question before last week.
Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to
1.0.8-BRIstuffed-0.2.0-RCh
the same problem occurs, but seems to be more seldom.
Attached is now the output of "zap show channel" .
-
I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM
equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.
2004 May 28
0
Problem with digits blending on inbound puls ed digits?
To answer my own question for the record:
The relevant timing parameters in zaptel.h are
#define ZT_MINPULSETIME (15 * 8) /* 15 ms minimum */
#define ZT_MAXPULSETIME (100 * 8) /* 150 ms maximum default, lowered
to 100ms */
#define ZT_PULSETIMEOUT ((ZT_MAXPULSETIME / 8) + 50)
And the pulse detecion loop that consumes these parameters begins at line
4866 of zaptel.c
The
2004 Jul 11
1
Echo issues (again...)
OK... so I'm not sure what I'm looking at. I've had the good old echo
problems on my Rev C FXO again this morning, so I thought I'd attempt
some debugging, though I'm not sure what I'm looking at.
This call has echo.
Channel: 2
File Descriptor: 20
Span: 1I>
Extension:
Dialing: no
Context: incoming
Caller ID string: "External Call" <99999999>
Destroy:
2005 Aug 12
3
TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a
TDM400P with (1) FXO card on port 4. Inbound calls are always successful
but outbound calls fail 75% of the time with intercept messages from my
dial tone provider that include "we're sorry, your call did not go
through", and "we're sorry, when placing a local call it is now
necessary to dial an area
2004 Jun 20
1
Data over Voice through Asterisk
Hi,
I'm trying to make a dialup internet connection through my asterisk
PBX. When I bipass the Asterisk box, I can get 51600bps. When I run
through the asterisk box, I'm limited to about 21600bps.
I have a TDM31B card.
Any help on speeding these connections up would be good - I was on the
understanding that if you bridged the channels, then the call should
essentially flow straight
2006 Apr 04
1
Ideal setup for PRI/T1 and TE110P
Hi all, I'm sure something similar has been discussed, but one can only
wade through the archives for so long.
I'm setting up a T1 and my telco has a bunch of questions it wants me to
answer. I know much of the TE110p is configurable to do any of this, but
I wanted to know if there is an optimal or preferred setup.
Any help would be appreciated. Here is the quiz my telco is giving me:
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 Apr 15
2
T1 Line install.. (UK Muppet)
Hi all, Muppet from the UK asking for help
We are just about to have a T1 line installed in our office in Dallas
and "Advantex" the supplier has sent a questionnaire asking a number of
questions. I have put the question area at the bottom of the email, we
will be using Digium's hardware. could anybody help :-)
In the UK when I asked for a E1, number of trunks required and the
2004 Jun 15
3
Outgoing DTMF when using BRI & i4l (Eicon Diva) - problems
Hello all,
This afternoon I had a BRI line installed by Telstra (our telco in
Australia). I'm using an Eicon Diva 2.02 PCI ISDN card with the isdn4linux
driver.
Incoming and outgoing calls with Asterisk work fine (and with no echo - my
main reason for getting ISDN). However, I can't seem to get outgoing DTMF
working (incoming works fine).
I made a call from my desk phone (Cisco 7940G)
2006 Apr 04
1
Ideal Setup for T1/PRI and TE110P - second try
Hi all, I'm sure something similar has been discussed, but one can only
wade through the archives for so long.
I'm setting up a T1 and my telco has a bunch of questions it wants me to
answer. I know much of the TE110p is configurable to do any of this, but
I wanted to know if there is an optimal or preferred setup.
Any help would be appreciated. Here is the quiz my telco is giving me:
2008 Oct 20
1
Zaptel FXO offhook when connected to PSTN
I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having
an annoying issue with the FXO ports. As soon as I plug either one into the
phone line it's as though the line is disconnected i.e. get disconnected
tone when trying to dial out, line is busy when dialling in.
The CLI shows the following:
trixbox1*CLI> zap show channel 4
Channel: 4
File Descriptor: 18
Span: 11*
2009 Aug 17
0
Echo on TE121B with hardware echo module
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
I recently upgraded my Asterisk system to Dahdi and now I have an echo
problem.<br>
<br>
I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium
TE121B PCI express card with a HARDWARE echo
2004 Jun 01
1
Zap and call pickup -- it don't work.
The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24
FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a
call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16.
*CLI> show version
Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux
The zapata.conf and extensions.conf are located here:
2010 Sep 21
1
digits in chan_dahdi
Hello
I use Asterisk with FXS extensions in chan_dahdi and I'm having trouble
detecting the digits in dahdi.
I dial 12345678, but only '16 'is received by the asterisk. The following
appears in the logs:
[Sep 21 18:11:44] DTMF [8536] channel.c: DTMF end '1 'received on DAHDI/10-1,
duration 0 ms
[Sep 21 18:11:44] DTMF [8536] channel.c: DTMF end accepted without begin
2007 Mar 22
1
Problem in using Two BRi Cards in Asterisk
Hi,
I have done my best and tired of searching the net about the problem. If anybody could help
would be a great favour.
Description of Problem
------------------------
I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on Trixbox 2 and aim
is to use in Asterisk as dailin and dialout. I compliled the driver as directed in the manufacture
manual. After installation dmesg
2007 Jun 28
1
Asterisk 1.4.5 Inserting Random Digits in Dialed Number!
Eeeeck! Asterisk is inserting random digits in dialed numbers.
So far I've seen it insert a 2 after the STD (area) code and insert an
extra 6 or 7 in the STD code. It's pretty repeatable although the
inserted number changes.
My Config is: Asterisk 1.4.5, Zaptel 1.4.3, Digium TE205P (rev 02).
There's an ISDN PBX on the second span and a BRI euroisdn on the first.
Calls from the
2003 Apr 25
1
Wait doesn't read DTMF? Was Re: Collecting dialed digits
A) Modify res_musiconhold.c and the application "WaitMusicOnHold" to
accept DTMF breakout
B) Create a call queue with a timeout of X and configure the DTMF
options properly. Then you can drop callers into this queue and effect
a music on hold for X seconds and allow DTMF breakout with no C code.
-----Original Message-----
From: asterisk@billheckel.com [mailto:asterisk@billheckel.com]