similar to: Using @Home 0.7 and wanting to debug dial plan problem

I'm new to this and have tried to find the answer in the discussions and docs but to no avail. I even read the posting saying the password issue has already been discussed. So, at he risk of being exiled, here goes. Question 1: I've installed 0.7 and can log into the asterisk server from windows by typing http://192.168.1.11 I can log in with wwwadmin and the password I set myself

Hi I have been searching the wiki and mailing lists and I cant see where my config is incorrect. I have a Digium tdm11b (1 fxo + 1 fxs) this is the output of cat /etc/zaptel.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0

Is there a configuration difference for clone X100P cards versus "compatible"? I have a similar problem to what David Shaw posted earlier today. 0.5 installed OK, but mine just with one X100P clone. Default config files, edited zapata.conf per the FAQs so it includes the line channel => 1 without the semicolon. Any outgoing call attempt returns "all circuits are busy"

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,

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

Does anyone know of a good solution to create a secure (encrypted) connection from a pocketpc (IPAQ 6515 in my case) to an asterisk server? Thanks Peter Kellner http://PeterKellner.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051012/2ee8b011/attachment.htm

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

Hello, I'm fairly new to asterisk and Trixbox, I'm setting up a Trixbox based email to fax gateway. At this time, I have a ZAP PRI link between the eFax server and my VoIPSwitch. The ZAP channels are configured, the B and D channels are up, and I have green link lights on either end of my cabling, but when I dial the number I have assigned to my eFax server, the call never seems to route

I've got a system where I'm integrating a Nortel Option 11c with a Trixbox 2.0.0 system using a Sangoma A101 T1 card. (Running on a Dell PowerEdge 350) We've got things mostly up and running and all seems well... except... If I call from a SIP extension (X-lite soft phone) dialing 9xxxx where xxxx is an extension on the Opt 11, the call goes through to the Opt 11 but I have terrible

I'm wanting both the voice and the configuration to be secure. (very secure). I don't care if it is SIP or IAX but I do need a softphone on the pocketpc I can use. I'd appreciate if you could take a look this weekend for me. Thanks, -Peter ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

hi i'm trying to configure a TE205P on asterisk@home i've edited /etc/sysconfig/zaptel adding this line: MODULES="$MODULES wct2xxp" now, when the system is loading, i can see that the wct2xxp module is loaded correctly but if i try the command: /usr/local/sbin/genzaptelconf i get: STOPPING ASTERISK STOPPING FOP SERVER Generating '/etc/zaptel.conf' Generating

Sorry about last posting, typo... I just added 2 Digium X100P cards. When my * box boots, it found them and configured them. When I enter genzaptelconf, it comes back with the following error: line 13: Unable to open master device '/dev/zap/ctl' Unloading zaptel hardware drivers: Removing zaptel module: rmmod: module zaptel is not loaded

I'm using two Rhino channel banks (first 12FXO/12FXS, second 24FXS). These connect to a Digium TE210P card. I'm running kernel 2.6.10 and I've tried Asterisk (w/zaptel) 1.0.9, 1.2 beta, and CVS from today. The results are the same for all versions: Right after I reboot, and modprobe wct4xxp, my analog phone connected to port 13 of the first channel bank (first FXS port) gets a dial

TRUNKMSD1=1 ; MSD digits to strip (usually 1 or 0) TRUNKMSD2=2 ; MSD digits to strip (usually 1 or 0) ; logn distance calls exten => _91NXXNXXXXXX,1,NoOp("Dialing: "${TRUNK}/${EXTEN:${TRUNKMSD1}}) exten => _91NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}}) exten => _91NXXNXXXXXX,3,Congestion When I dial

Hello, I'm trying to upgrade an Asterisk 1.2 linux box to Asterisk 1.4. I installed the following -rw-r--r-- 1 root root 10908541 Sep 21 13:25 asterisk-1.4.0- beta2.tar.gz -rw-r--r-- 1 root root 993921 Sep 21 13:25 asterisk-addons-1.4.0- beta1.tar.gz -rw-r--r-- 1 root root 80019 Sep 21 13:25 libpri-1.4.0-beta1.tar.gz -rw-r--r-- 1 root root 1523413 Sep 21 13:25

Hi, I am having problems with an Asterisk with a Digium TE205P card. The issue is that the Ring Back Tone is noisy. I am making modem's calls and this noise influences on the initial negotiation protocol, so modems have to recall. My configuration is: Asterisk version: Asterisk 1.4.21.2 Linux version: CentOS release 5.2 (Final) Card: Digium TE205P

Greetings and I thank you in-advance, I have a system installed with both Trixbox 1.x and 2.x installed and I am trying to utilize a wildcard x100p but it does seem hang-up or release the line. I understand that the Wildcards are not recommended, but I need to get a test system up for evaluation. Here are the steps followed: Run genzaptelconf and red alarm appears in zttool Run again and it

>From the command prompt type: help-aah This will give you a list of commands to change passwords. For example: Commands Descriptions ----------------------------------------------------------------------- config set the local time zone and keyboard type netconfig configure ethernet interface genzaptelconf autoconfig Zaptel

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

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