similar to: TE110P - Asterisk@Home Install Problems - Televantage 3 T1

Displaying 20 results from an estimated 1000 matches similar to: "TE110P - Asterisk@Home Install Problems - Televantage 3 T1"

2005 Sep 14
1
TE110P - Asterisk@Home Install Problems
I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single asterisk@home system with a single T1 card. Robbed Bit T1 ami, d4. ------------------inbound call
2007 Sep 24
0
Asterisk Dropping Calls
Hello, I am having an issue whereby calls are being dropped randomly. I have an ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk install is based on Trixbox 2.0. However, I have updated the source code to the following. The Asterisk release is asterisk-1.2.20. Zaptel release is zaptel-1.2.18. And libpri release is libpri-1.2.4. I have include an extract from the Asterisk log
2007 Sep 30
0
Asterisk Dropping Calls (Richard Young)
> > Hi, Remove usecallingpres=yes busydetect=yes from your zapata.conf file. and the restart asterisk. Hopefully you will not faced drop call issues. Regards, Vidura Senadeera. Message: 3 > Date: Mon, 24 Sep 2007 12:29:40 +0100 > From: "Richard Young" <Richard.Young at intrintech.com> > Subject: [asterisk-users] Asterisk Dropping Calls > To:
2004 Aug 02
1
Asterisk as Front-End for Artisoft Televantage 6
Anyone had experience 'marrying' the two? We had setup * to front end Artisoft's Televantage. It works with some issues need to be resolved: - Inbound calls could not properly handled and routed by Televantage's Call Classifier. It goes directly to the Televantage's default auto attendant. Correct digits not being passed by * to Televantege? - The caller's Caller ID does
2004 Nov 19
0
TeleVantage client on Wine
I am attempting to install the <a href="http://televantage.activelogic.com">TeleVantage Client</a> into wine. I have compiled Wine-20041019 from source. Upon running the installer, I get way too many errors: alexander@reservations:~$ wine .wine/drive_c/AppsPatches/netsetup/setup.exe fixme:msi:MsiGetProductInfoA "{8715D03E-39E4-4260-BB83-C63CA64D7660}"
2005 Aug 02
0
Televantage 4
Hello, I'm not sure if anyone can help, but I figured I'd ask anyway. I am attempting to run the Televantage 4 Client under Wine 20041019 with WineTools 2.1.2. It installs fine, but it does not run. If I make Wine emulate win98, I get this error: fixme:vxd:VXD_Open Unknown/unsupported VxD L"secprov.vxd". Try setting Windows version to 'nt40' or 'win31'.
2008 Jul 17
0
TeleVantage Call Monitor & Asterisk
Hello, Have any of you had the chance to see what TeleVantage call monitor is? It basically shows queues, other extensions, calls in waiting, voicemail, etc. Does anyone have if there is something out there like it, so each and every agent can have one of those? And not just a manager? I'll be putting in a screenshot if someone wants to see. Thank you, Mark --------------
2010 Dec 10
1
TeleVantage Client 8
Hello. I am trying to run TeleVantage Client 8.00.5556 on Wine but it doesn't work all the way. I have done the following, after some tips here and there: Wine 1.2 under Ubuntu 10.04 in a VMware Player as I am testing. OS version set to Windows 7 Server is running on Windows 2003 on the LAN workgroup and has windows clients today (something I had hoped to change) Client installed through
2005 Jul 05
1
Newbie question reg. Asterisk and Channel Access Bank I and TE110p
Hi, I have some problem to get this setup working. I have a CAC Channel Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2) and I have a TE110p installed in this box. What I want to do is, just to be able to dial one of those lines that already are connected to the channel bank, and transfer that call through TE110p and Asterisk to a user agent somewhere through Internet.
2005 Sep 19
1
problems with PRI
Hi, I configured an asterisk box with 1 Digium Wildcard TE110P T1/E1 Card 0 I setup the jumper in e1 position. my zaptel.conf : defaultzone=it loadzone=it #gestione PRI span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15 # set this to 1-15,17-31 for E1 dchan=16 # set this to 16 for E1 bchan=17-31 # set this to 1-15,17-31 for E1 # Asterisk starts correctly, I see th 30 channels. Anyway I cannot put
2009 Jan 12
2
a zaptel problem
Hello, I have a problem with zaptel. I hope you can help me. I installed and configure zaptel. ZAPTEL.CONF span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = de defaultzone=de But the output of cat /proc/zaptel/* Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) HDB3/CCS/CRC4 RECOVERING 1 WCT1/0/1 Clear (In use) RED
2005 Sep 16
0
Zap failed
I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Asterisk 1.0.9, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com> .... [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Sep 16 20:36:51 ERROR[6750]: chan_zap.c:6246 mkintf: Unable to get parameters Sep 16 20:36:51 ERROR[6750]:
2006 Mar 12
1
interop problem: "Missing handling for mandatory IE 24 (cs0, Channel Identification)"
Hi everybody, I've connected Asterisk 1.2.5 (libpri 1.2.2, zaptel 1.2.4, Linux 2.6.13.2) to an Avaya-Tenovis PBX via a PRI/E1-line. Calls from SIP-phones via * to the PBX work fine. However, incoming calls to * only result in: -- XXX Missing handling for mandatory IE 24 (cs0, Channel Identification) XXX -- which seems to be an * problem, because a Windows-fax-machine works fine on a PRI
2008 Feb 20
0
Unable to create channel of type 'Zap' with ecmg2 and kernel 2.6.23
Hi, I have a working Asterisk 1.2 server on kernel 2.6.22 with the OSLEC echo canceller on a Digium PRI card. I recently switched to kernel 2.6.23 with the MG2 echo canceller (nothing else changed). Each time I try to establish a call on the PRI line I get a congestion signal. in /var/log/asterisk/full: Feb 20 08:09:43 VERBOSE[10657] logger.c: -- Executing
2004 Aug 04
4
Problems with E100P
Hi, I'm having trouble configuring and E1 link , I know the E1 is a PRI and the switchtype is 5ess. The problem is that everytime I try to dial I got and error saying that it was unable to open the zap channel . Devices are created in the /dev/zap directory and I can open them with a cat /dev/zap/1. I can also see the channels in /proc/zaptel/1. Thanks in advance [zaptel.conf]
2005 Aug 05
1
Problem running Astrerisk after defined card in /etc/asterisk/zapatal.conf
I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15' Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered
2006 May 23
1
chan_zap.so error, asterisk stopped
Hi!I've a problem with asterisk,it doesn't start..it's stopped..I used the amportal start command and this is the result: SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk.
2006 Jul 18
1
SpeexEncoder requires 320 samples to process a Frame, not 160
Hi guys I have tried compiling this attached code, I made all the buffers 320, there is no trace of a 160 buffer, but I get a " SpeexEncoder requires 320 samples to process a Frame, not 160" error. Maybe there's something I'm missing, here's my code: import java.io.IOException; import java.io.FileOutputStream; import java.io.File; import
2006 Mar 31
1
Asterisk, QSIG and Tenovis PBX?
Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. Calling from a Tenovis phone to a SIP phone (i.e. traditional phone -> Tenovis PBX -> QSIG -> Asterisk -> SIP phone) works with the following messages: --- Don't know what to do if second ROSE component is of
2011 Feb 14
1
Asterisk Call File using Local Channel not passing Variable back to Dialplan
Hi Everyone, I am trying to pass a variable using the .call files but it turns out blank. Can someone please point out what might be wrong here: */tmp/spool-file.sh* *----------------------------------------------------------------------* echo "Channel: Local/s at callback_leg*1*/n CallerID: \"Call-back\" <123456> MaxRetries: 0 RetryTime: 10 WaitTime: 45 Context: