Displaying 20 results from an estimated 1000 matches similar to: "FW: zapata.conf: recent changes?"
2006 Jun 21
4
zapata.conf: recent changes?
Hi,
after a few of upgrades, I noticed these messages in full debug log:
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring
2011 Feb 18
1
Asterisk with TE 121 DADHI incoming calls fail
Dear Customer Support,
i connected the asterisk to a e1 interface of our hipath4000. outgoing
calls from a sip peer of my asterisk to an up0 telephone which iss
connected to the hipath4000 are working. If you want to dial from an up0
device to the e1 interface where asterisk is connected to, you have to
use the prefix 83. But when you enter the 3rd cipher this error appears
at the cli
2006 Jan 14
1
Problem with just one number!
I have this setup:
(PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones
and a few of VoIP phones directly connected to Asterisk.
Calling a number (just one!) - an automatic responder (IVR) -
from VoIP phones works, from analog phones doesn't work:
NOANSWER after a few seconds.
I'm using no 'r' in dial options (this caused a problem with an IVR some
time
2006 Jan 16
2
Problem with calls starting from a legacy PBX
Hi,
I have this setup:
E1 PRI PSTN -- Asterisk -- Alcatel PBX - analog phones
Can someone tell me what's wrong with this call initiating from an analog
phone connected to Alcatel PBX?
It dies with NOANSWER but all works if I call other destination numbers.
Dialplan is a simple Dial(zap/g1/0984465691) statement.
At the end you'll find also zapata.conf.
2008 Feb 18
1
PRI dialplan/prefix
hi.
could somebody explain how exactly the following parameters
in zapata.conf work:
pridialplan
prilocaldialplan
internationalprefix
nationalprefix
localprefix
privateprefix
unknownprefix
the wiki & comments doesn't quite explain them. and
phone companies are absolutely no help.
i've setup systems in the US & China with trial & error
until it works. now i'm setting up a
2006 May 09
1
PRI in Shanghai China
hi folks.
does any one have experience setting up E1 PRI in Shanghai, China?
it works fine when we use SIP phone to dial out, however when
using forward function on the same phone, it seems like it's dialing
out but there's actually no respond from the phone company (China Telecom)
and eventually the dial command will timed out.
here's our PRI portion of zapata.conf:
2009 Apr 03
1
ISDN Timer T309
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
Hi everione,<br>
<br>
I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1,
libpri 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my
tests, the timer fail with a telco
2011 May 17
0
Type of number in outgoing SETUP frame
Hello, all.
Could you tell me how to set the type of number in the outgoing SETUP
message sent over PRI trunk?
I need to have:
Called Number (len=18) [ Ext: 1 TON: Unknown Number Type (0) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '318989263037666' ]
But always have:
Called Number (len=18) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan
2006 May 05
2
AW: AW: DTMF detection when outgoing call tomobilephones
Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2
I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf.
The detection is not working with call file, manager originate and not with the dial command to the mobile.
I have no ideas left.
I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...)
But with the same vaules on a second call there
2010 Feb 18
0
ISDN phone not ringing. ISDN PBX not answering?!
Hi,
I've set up an Asterisk as voip gatway:
VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx.
Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset.
I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode.
The msn is set at the dect phone/base station
2008 Jun 20
1
FXS port doesn't provide dialtone
Hello everyone,
I want to connect a fax to an FXS port (TDM420P). For testing purposes,
I connected an analogue phone to it first. However, when I pick it up, I
cannot hear anything at all.
The power cable is plugged into the card, the port is configured to use
fxo-signalling. Also, immediate=no. Here's the files:
/etc/zaptel.conf:
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand
2007 Jul 26
1
tdm400p fxs module busy
Dear All
The setup is te110p with an 8 channels PRI to make and receive all calls.
SIP phones throughout the company.
TDM400p with 4 FXS modules to send/receive faxes and make credit card
transactions.
I have an analogue phone on the tdm400p for testing.
I can receive calls to the exten. There is a dialing tone.
However, when I try to make a call I get a busy signal.
Asterisk stated busy then
2010 Sep 15
1
One way audio when overlapdial is set to yes
Hi Group,
I am currently facing a dead end and any help will be much appreciated.
I have an a104d installed in an asterisk box, two of which is configured on ISDN
pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am
getting one way audio when a local on the hipath tries to make a pstn call but
no issue on incoming calls from pstn going to the hipath locals.
local
2007 Jul 12
0
No subject
the CNAM info in the Q.931 call setup message. I've tried all
permutations of switchtype (dms100 & national) and facilityenable that I
can think of, but I still don't see CNAM coming out the other side.
Telco confirms that "Name Out" is enabled on our PRI.
Any pointers on what I'm missing, and/or how to debug further?
zapata.conf:
---
[channels]
context=default
2007 Jul 12
0
No subject
the CNAM info in the Q.931 call setup message. I've tried all
permutations of switchtype (dms100 & national) and facilityenable that I
can think of, but I still don't see CNAM coming out the other side.
Telco confirms that "Name Out" is enabled on our PRI.
Any pointers on what I'm missing, and/or how to debug further?
zapata.conf:
---
[channels]
context=3Ddefault
2007 Nov 18
0
facilityenable in zapata.conf
Can someone explain what the facilityenable setting does in zapata.conf
I've read the wiki & archive, but it's not even clear what an ISDN
"facility" is.
Thanks,
MD
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2009 Jul 03
0
DAHDI CDR problem
Hello gang,
We just got MaBell to turn on our callerid. I tested the
capability with a southwest bell box and a plain phone, so I know the line
is sending the signal. I'm running Asterisk SVN-branch-1.4-r204834 using a
TDM400P card. Here is my dahdi_cfg -vv output:
dahdi_cfg -vv
DAHDI Tools Version - 2.1.0.2
DAHDI Version: 2.1.0.3
Echo Canceller(s): MG2
2009 May 23
1
1.6.0.9: Unknown signalling method 'pri_cpe' ??
I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card.
I can't make any connection over the T1.
From CLI:
ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling
method 'pri_cpe' at line 37.
cat chan_dahdi.conf
cat chan_dahdi.conf
[trunkgroups]
[channels]
language=en
;internationalprefix = 00
;nationalprefix = 0
context=from-pstn
switchtype=national
2006 Apr 28
1
Official TE411P echo settings??
I have seen conflicting references in regards to the Digium Wildcard TE411P echo settings in zapata.conf.
Does anyone have the official word on this?
Should echo cancel be enabled in zapata.conf if the card has built in EC?
If so, should a particular EC method be compiled into the zaptel build?
My reference, which has echo:
My zaptel is 1.2.5
context=from-pstn
switchtype=national
2006 Apr 04
0
some problems with asterisk and E1
Hi,
I am using asterisk 1.2.5 and have some problems with asterisk connected with
an E1 card to our PRI. Dialling in and out generally works. When someone dials
in from a mobile phone, all numbers are sent as a block, and the called
extension rings as intended. when someone picks up his phone handset, waits
for a dial tone, and then dials in manually, the call will be redirected to
the