similar to: Fast busy signal

Hi All, I've installed asterisk and manually configured IAX/SIP users. Everything works fine, I'm able to call other extensions. But when I installed AMP and created new extensions, I'm not able to call those extensions. I get the message that the extension is busy and it is forwarded to voicemail. What am I missing here? The workaround I found is by modifying the

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

I have fixed this before, but I cannot for the life of me remember how I did it. I have a TDM400P with one fxo module and one fxs module. I setup Asterisk @Home and everything works fine, except when I try and call out. If I call out with a SIP phone it calls the zap extension and not the pstn line? If I take the zap extension offhook and call with the SIP phone it dials out the pstn line

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

Hi, I am thinking of purchasing a cheap Dlink VPN for testing purposes for use with my Asterisk box and would like to ask the list for advice on how to pick a VPN that will work with my box. I am a newbie to both VPN's and Asterisk so any advice will be appreciated. Thanks, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi- Can someone help me conceptualize how I could setup an * box which features both English and Spanish prompts? I know it's possible to configure the box to use one or the other, but how could I setup a multi-lingual box? Any suggestions? Thanks in advance, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Sherwood, Your intentions are noble and your desire to build this, fullfills an immediate need for business. If your intention is just to build a GUI for Asterisk, read no further. If your desire is to build something more purposeful, your best bet would be to see the existing commercial GUI/HostedPBX offerings like Pbxware and Switchware from bicomsystems.com ( http://www.bicomsystems.com)

Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2)

I've spent all day today trying to fix an echo problem and I've made no ground whatsoever. I have Digium TDM400 with 3 FXO & 1 FXS. I've tried this computer at two completely different sites with different phone providers. I've tried compiling & installing different versions of Zaptel (currently running 1.4.0, started at 1.2.9 and worked my way up). I've tried

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

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

Hi, Is it possible to setup an Asterisk system that can allow someone to dial in using a DID and listen to their e-mail? Has anyone done this? Thanks, Mike C.

After tinkering with a hand-knitted extensions.conf based on "Asterisk - TFOT", I've now set up a server with Asterisk@Home and am experimenting with it. I'd appreciated any advice from the more experienced list members about which way to proceed. We (my wife and I) have two separate micro-businesses with two POTS lines plus fax. I'd like to have inbound calls on the two

http://pastebin.com/raw.php?i=rykHdvBh bix.hu and www.yahoo.com are "pingable" test sites. 127.0.0.1 could not be pinged [firewall drops all icmp] i have a "oneliner" that echoes if theres "internet connection or no". $ ping -W 1 -c 2 bix.hu >& /dev/null && ping -W 1 -c 2 www.yahoo.com >& /dev/null &&

Hello, I have 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my <mailto:*@home> *@home 2.8 running on top of CentOS. Both FXO Ports are on ONE Digium card. What I would like is: If someone calls extn 281 on my Alcatel PBX it is routed through to Extn 233 on my * thruogh FXO port/module 4 If someone calls extn 282 on my Alcatel PBX it is routed through to Extn 234 on my *

Hello, I have another little problem with my ZAPs channels, in fact, when I received a call, I heard no sound while in the CLI, sound is played: -- Starting simple switch on 'Zap/4-1' -- Executing [s at from-zaptel:1] Answer("Zap/4-1", "") in new stack -- Executing [s at from-zaptel:2] BackGround("Zap/4-1", "hello-world") in new stack --

Christoph, I understand this is the better option if HVM is available.. Im not sure how to make use of this kernel in a non-HVM, PV environment. I'm a little disturbed by the fact that there isnt PV support in CentOS 8, I hope it comes later so at least PVH becomes an option. As for using this kernel in a PV only environment, the only route would be attempting an in place upgrade and try to

Hello together, I am an absolute noob in R and therefore I need help urgently. I have received a script from my tutor with plot functions in it. However, I can' manage to adapt these plots. The hole script is as follows: setwd("E:/") ##### (1) Read data ### dat <- read.table("Komfort_Tatsaechliche_ID_Versuchsreihe_1.txt", header=TRUE, sep="\t",

Hi I have a system with one asterisk box in front and a few PSTN gateways in the back. When a call comes from PSTN, it's directly forwarded to the edge/user asterisk box. Now if a number is dialled, and that number is not in use, blocked due to lack of paying etc, I want to signal that back to the PSTN gateway, making this playback or playtones to avoid picking up the phone. Can someone