Displaying 20 results from an estimated 2000 matches similar to: "The sound is played but I did not hear"
2005 Sep 28
2
Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unkn own signalling method 'pri_net'
Did you compile and install libpri *before* Asterisk? I had same problem
(among others) b/c I didn't install in the correct order. Try the awesome
asterisk_update.sh shell script.
Are you trying to emulate CPE or NET? Try signalling=pri_cpe
Check for whitespace behind the statement, zapata.conf seems bitchy about
whitespace.
hth
-----Original Message-----
From: Steve Totaro
2005 Sep 28
1
Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net'
Any ideas?
51] logger.c: [chan_zap.so] => (Zapata Telephony)
Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found
Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing
2006 Jun 06
5
HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.
Hi all,
I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in
my office. the out going calls symptom like when called party pickup the
phone but the calling party still hearing the ring tone from the IP phone.
Please light me up. it been many sleepless night by googling around
trying to get the right answers.
The digium card running on Intel 915G chipset. Below are my zaptel
2005 Jul 17
2
HFC BRIstuff woes
Hi All,
It's broken !! (drat)
Asterisk if failing to load with the following error (taken from end of
/var/log/asterisk/full) after adding bristuff.
Can anyone help please?
Jul 17 19:57:54 VERBOSE[2503]: == Registered channel type 'Phone'
(Standard Linux Telephony API Driver)
Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so]Jul 17 19:57:54
VERBOSE[2503]: [chan_zap.so] =>
2005 Jun 07
2
Help! Zap echo on bridged calls
I've been going nuts lately trying to get rid of an annoying echo
problem that makes my asterisk server unusable when clients try to call me.
Here's the breakdown of the issue - Hoping that someone can throw me a
clue:
My setup is as such:
Single AMD Athon machine with X100P clone card and voip through multiple
providers .
* Inbound calls through the X100P that do not bridge to
2005 May 20
1
MFC&R2 Venezuela with libunicall
Hi,
I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code.
All libs compiled successfully and the E1 have a green light!
I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working.
My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P
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
2007 Aug 09
2
Terrible clicking on T1
Hey All,
I have an Asterisk box connected to a Nortel Option 11C via a T1. In the
Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI
card. The Nortel is also hooked to the PSTN via a T1 on a different
NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files
below.
Our issue is that when a call is sent over the tie line between the two
systems, the audio on the
2008 Mar 27
3
problem about voice when using TDM2400p with VPMADT032 echo canceller module
hi you,
I'm having problem with voice quality on my trixbox using TDM2400B.The trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'background crackle/buzz' coming back when they talk.
anyone have the same problem? pls help me. thanks a lot.
my trixbox and config
2006 May 27
3
TDM
The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk
1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do
the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point
that the cable plugs into the card.
Here is my /etc/zaptel.conf
loadzone=us
fxsks=1
and here is my /etc/Zapata.conf
[channels]
language=en
#include
2007 Mar 15
2
A200 card problem
Hi -
I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't
make it work- currently, asterisk will not startup because of a bad
module. Below are some log files/config files. If anyone has any
suggestions, I'd appreciate it.
I used Trixbox 2.0 and followed instructions on (http://
sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems
running through or
2007 Jun 26
0
No CID on Zaps - TDM400
I'm running Trixbox 1.2.3 with 2 TDM400s (FXOs).
With Trixbox out of the mix and a regular phone connected I get the CID
fine yet Trixbox shows 'unknown':
dialparties.agi: Caller ID name is 'unknown' number is 'unknown'
dialparties.agi: Methodology of ring is 'ringall'
Here is my Zapata.conf if it helps:
#############################
;
; Zapata telephony
2011 Jan 24
2
Outgoing FXO calls have no audio with callprogress=no
My outgoing FXO calls are answered but have no audio in either direction
if I have callprogress=no in chan_dahdi.conf. If I change to
callprogress=yes then the audio returns. My chan_dahdi.conf file is
listed below. Can anyone point-out why callprogress=no isn't working?
#cat /tmp/a
[trunkgroups]
[channels]
language=en
context=incoming
toneduration=40
;usedistinctiveringdetection=yes
2011 Jan 03
1
Clarification on DAHDI Fax Detection
Hi folks,
I was hoping that someone might be able to help clarify some confusion I
have on DAHDI Fax detection after spending some time searching. My
understanding is this:
1.) Echo cancellation is automatically disabled upon recognition of a
CNG tone, regardless of the faxdetect setting. This can only be disabled
at compile time.
2.) faxdetect=incoming will, upon detection of a CNG tone,
2015 Feb 18
2
Res_fax - FAXOPT(faxdetect)
I solved the issue by not answering the call as I assume others have done.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Administrator TOOTAI
Sent: Wednesday, February 18, 2015 12:50 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect)
Hello
Le
2013 Oct 30
1
dahdi fax catch-22
I've got a Digium Wildcard TDM410P with one POTS line and three
extensions. One of the extensions is connected to a fax modem. This kind
of works, but there's a gotcha. If I set faxdetect=incoming in
chan_dahdi.conf, then incoming faxes do get routed to the modem and this
all works, but outbound faxes fail, with a message like this:
[Oct 29 21:57:09] WARNING[16732][C-00000000] app_dial.c:
2006 Jun 19
5
faxdetect questions - Please HELP!
I'm using IAXmodem and Hylafax with 'faxdetect=incoming' and things mostly
work pretty well. My main lines come in via T1 DID. Today, HR got tired of
having someone read and forward their faxes to them and requested we bring
their physical machine back on line. I have been able to get the fax
forwarded to the appropriate zap channel, but I cannot seem to get it to
stop
2007 Jul 24
1
[beginner] Problem of detecting call
Hello,
I have some problem to start asterisk.
First I have followed a lot of tutorials to complete correctly the install
process. Now it works when I type zttool I can see when I am or not
connected to the PSTN.
But, I run asterisk with vvvv verbose and I can't see the call detection.
There is no detection of the call.
I have a X100P card FXO with only one line. So only one channel
I
2015 Feb 17
2
Res_fax - FAXOPT(faxdetect)
Hi,
as stated in the documentation, it's allowed to set
FAXOPT(faxdetect)=yes/no to allow fax detection.
It's done (see below) but still fax detection :-( Extension 300 is
hylafax with iaxmodem.
On the upper Asterisk gw it's the same, despite the faxdetect set to no
we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile
phone calling the 0123456789 PSTN number.
2006 Feb 09
1
clid and src fields wrong in cdr
Hi all,
I have a strange problem, regarding zap channels and cdr.
I am using asterisk bristuffed version
Asterisk 1.2.2-BRIstuffed-0.3.0-PRE-1i, Copyright (C) 1999 - 2006 Digium,
Inc. and others.
with two billion ISDN cards. I also installed asterisk addons, last stable
version via cvs
internal calls, or calls starting from internal sip or iax phone are
recorded in the cdr all without any