Displaying 20 results from an estimated 2000 matches similar to: ""Unable to get parameters" while configuring FXO cards, any ideas?"
2003 Dec 28
0
Is there something wrong with "show manager commands"?
Is it just my box, or is there something flaky in the implementation of
"show manager commands"?
Note: I'm using putty. About half way through this, I toggled my KVM over to
the desktop and logged in to try and recreate it. The output was the same as
the last two entries in this dump.
bebop*CLI> show manager commands
bebop*CLPing Ping
bebop*CLLogoff Logoff Manager
2004 Apr 04
0
Strangeness
Hi,
I restarted my * box before and now my second X100P card has stopped
showing up in Asterisk. It shows up in my dmesg in Gentoo, but only 1
channel is being registered:
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, FXS Kewlstart signalling
== Registered channel type 'Zap' (Zapata Telephony
2005 Mar 16
0
Problems with TDM400P and asterisk on Linux 2.6
Hi there,
I have a TDM400P (1 x fxs, 1 x fxo) which I'm attempting to run on linux
2.6 (gentoo), without much success at the moment. I have previously had
it working on a 2.4 installation, but when I moved to a new box and
installed a 2.6-based system, it failed to work. In both cases I'm using
whatever (libpri, zaptel, asterisk) is checked out by default (I assume
that means HEAD)
2005 Sep 03
1
newbie install problem. And I already searched everywhere!
Hi,
When startin * it tells me:
================================
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf: Updated
conferencing on 1, with 0 conference users
-- Registered channel 1, PRI Signalling signalling
Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf:
2004 Dec 03
0
ZAPHFC, Asterisk does not load with signalling = bri_net_ptmp
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
Hi,<br>
<br>
I am trying to get zaphfc running in nt mode with asterisk.
2004 Jan 31
2
TE410P E1 PRI problem
Hi everyone!
Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a
2005 Aug 08
4
TE110P flashing red/green when PRI connected
Hi
I'm having difficulties getting up my TE110P (running as a E1) when I connect it to the PRI. If I start the server with a loopback connector everything seems fine and the led is green but when I connect it to the PRI the flashing starts ....
I can't seem to find anything in the log that suggests what could be wrong
.....Aug 8 11:33:00 DEBUG[1369]: Updated conferencing on 31,
2005 Feb 18
0
Installing Asterisk on Mandrake 10.1 Official
I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too.
The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was
2005 Jul 27
0
Sending DTMF Tones Offhook
Greetings All!
The Asterisk Call Manager works great. But I have one question for
anyone who has used it. I cannot get the system to send some DTMF
tones down the channel once the call has been made. Below is the
script I am using to make the call, and start recording the channel.
I am starting to make a system the will use asterisk to become an
automatic random quality monitoring system
2005 Jun 15
1
app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP -> SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give
2005 Sep 07
1
Several SIP clients behind router register with the same IP, messing up call routing, any ideas?
Dear all,
Has anyone seen this before and can suggest a solution?
I've got 3 sip clients behind the router, and they all register with
Asterisk using the same IP address.
Now, wenn all are registered, all the calls get routed to the client that
registered most recently, but not to the correct client.
Also, there seems to be some problems registering all 3 clients
simultaneously.
Could
2005 Sep 08
1
Additional: Several SIP clients behind router registerwiththe same IP, messing up call routing, any ideas?
Hello,
I'm still looking for any ideas on this problem:
I've got 3 sip clients behind the router, and they all register with
Asterisk using the same IP address.
Now, wenn all are registered, all the calls get routed to the client that
registered most recently, but not to the correct client.
Also, there seems to be some problems registering all 3 clients
simultaneously.
I have
2003 Nov 19
1
FXO card still won't pick up...
I recently updated (fresh checkout) to the newest zaptel and Asterisk.
The one I was using before was a couple of months old.
After updating, my zap channels don't work. They won't pick up incoming
calls or dial out. When I try to dial out I get:
-- Executing Dial("SIP/3064-564c", "Zap/g1/ww954.......") in new stack
NOTICE[245776]: File app_dial.c, Line 698
2006 May 09
2
Problems with TDM400P and FXO modules
Hi , I have an asterix (1.2.7.1) box with 3 TDM400P and 10 FXO modules, and
zaptel (1.2.5).
The problem is that 6 modules can't be initaized and i get the folowing
error:
May 7 09:17:37 WARNING[7739] chan_zap.c: Unable to specify channel 9: No
such device
May 7 09:17:37 ERROR[7739] chan_zap.c: Unable to open channel 9: No such
device
here = 0, tmp->channel = 9, channel = 9
ztcfg output
2008 Mar 18
1
Sangoma FXO/FXS config
Hi all,
I bought a Sangoma A200 card from an online supplier and explained
exactly what I wanted,
3 incoming phone lines to PBX and a life line (some where to connect a
standard BT phone to the PBX incase the power goes, making the BT
phone ring).
I was told to order
1 x FXS module (2 FXS ports)
2 x FXO modules (4 FXO ports)
However being a complete noob, I have connected the 3 lines to the
2009 Feb 24
2
Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card
Hi I have been having a rough time getting a Sangoma A200/Remora FXO/
FXS Analog AFT card set up properly.
The main issue is that the card has four ports and as far as I can
tell Asterisk is only seeing two. On the two that it recognizes the
"Green" FXS ports are not green, they just are not lit. The "RED" FXO
ports are indeed red, but from what I have read your not
2005 Jul 22
1
Problem with Zaptel FXO..
Hi all, i've installed AMP and Asterisk following the INSTALL file and i have
a problem with the TDM04B with 4 FXO:
[root@srvoip ~]# ztcfg -vv
Zaptel Configuration
======================
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Kewlstart (Default)
2003 Dec 03
2
Re-routing of existing calls
Does Asterisk have the capability to re-route calls that have already been
connected?
By this, I mean:
1. A call comes in to Asterisk.
2. It is routed to an extension as normal.
3. This extension answers, and the conversation starts.
4. After a few minutes, a plugin that I am writing decides that it wants to
transfer the call to somewhere else.
5. It signals this to the core of Asterisk (this
2004 Sep 01
1
FXO Disconnect supervision
Does anyone have disconnect supervision working on their TDM400P or X101P
cards (to/from the telco)?
Googling "telco disconnect supervision," I found the following description:
http://mirror.lcs.mit.edu/telecom-archives/telecom-archives/TELECOM_Digest_Online/1559.html
[cut]
Re: How Do I Get "Kewlstart" From my Phone Company?
Paul A Lee
Tue, 24 Aug 2004 16:58:16 -0400
...
It
2006 Mar 27
2
FXO without answer supervision
Simple question that google hasn't helped much with (likely poor search
terms)
I just installed a Sangoma A200 with overall good results. Initial
tests with both
incoming and outgoing calls were very positive. Until I made a normal
call that
lasted more than 30 seconds.
I setup the FXO with kewlstart signalling, and the outgoing call is
never registered
as 'Answered' and the dial