similar to: asterisk-gui installation hangs

Hi All, This is my first post. I searched the archives and found something similar and I tried some of those suggestions. I changed the file permissions on the scripts directory to 777 (which doesn't seem secure), I also manually ran the detectdahdi.sh script. The response is "None". I am running Mac OS X 10.5.7 with Asterisk 1.4.26.1 which I compiled from source. The Asterisk Gui

Hello, I've installed Asterisk and Asterisk GUI 2.0. The GUI says "No Analog Card found" and /etc/asterisk/ztscan.conf is empty. I see the following message from asterisk, <snip> -- Executing [executecommand at asterisk_guitools:1] System("Local/executecommand at asterisk_guitools-a145,2", "uptime >

When I load the asterisk 1.4 gui and log into "/asterisk/static/config/setup/install.html", it tells me "No Analog ports has been detected on your system". I have 2 Wildcard X100P cards that are properly installed. Ztcfg shows no problems. I also get the following message from the asterisk console " app_system.c:105 system_exec_helper: Unable to execute

Hi, I am trying to setup a simple home voip service w/ * I have compiled and installed the svn source as a first step I am trying to configure SIP for inside my network. I have a handful of softphones and a few hardphones that I want to all be able to call each other I have configured users.conf with a single softphone(kphone) and have tried calling itself (ext 6000) and the demo from the

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

Hi, Is it possible to monitor conversation of logged in Agents? Currently I am using ZapScan to monitor incoming calls, but I would like to monitor individual agents. raj

Anyone successfully implemented a solution for allowing ZapBarge call monitoring only for a specific group of agents calls? The issue I see is that the feature only works on zap channels, and all of the agents (in many cases) are IP phones. Allowing ZapBarge and ZapScan on the TDM PSTN (t100p) interface has privacy issues for senior managers, but would allow all outbound zap calls to be

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091111/2b828eff/attachment.htm -------------- next part -------------- Hello. I am trying to execute an fax reception script and i am getting the following: [Nov 11 08:40:52] WARNING[12800]: app_system.c:88 system_exec_helper: Unable to execute '/var/lib/asterisk/scripts/mailfax ""

From: Malcom Kemp Sent: Wednesday, April 25, 2007 11:19 AM To: 'asterisk-users@lists.digium.com' Subject: AsteriskNOW generation of zapata.conf file I am a new user of Asterisk, and an trying to use AsteriskNOW. The test system is dual processor, so I am using the beta 4 version. I am currently trying to manually configure, as the GUI does not seem to let me accomplish what I need.

Hi Timo, hi all! Today i noticed imap throws segmentation faults and dumps cores. I looked into logs I can see: 2013-03-07T12:12:52.257986+01:00 meteor dovecot: imap(marcinxxx at kolekcja.mejor.pl) <7sRXtFPXYAA+eX93>: Error: Corrupted dbox file /dane/domeny/mejor.pl/mail/marcin//.mdbox/mailinglists/storage/m.75 (around offset=2779212): EOF reading msg header (got 0/30 bytes)

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

Hello, Let me start by saying I have checked the wiki and the archives and did find some relative information. I tried the suggestions in those threads, but still have the same problem. I'm using the CVS Asterisk from July 11, 2005. Redhat FC2 SpanDSP 0.0.2pre18 Libtiff 3.5.7 Digium PCI card 1 FXO, 1FXS. I have a single POTS line coming, but I have 2 numbers and am using

Hi Folks, I've been searching for an specific feature on asterisk and I found an e-mail from "John Todd" asking for the same thing. http://lists.digium.com/pipermail/asterisk-users/2004-May/045882.html To be able to listen (zapbarge, zapscan or chanspy) an specific channel and can talk to one side (the operator). That feature is very usefull in call centers in Brazil so if you want

Hi, You can try ChanSpy http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy. Idris _____ From: phil.dawson@marnock.com [mailto:phil.dawson@marnock.com] Sent: Wednesday, June 21, 2006 12:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Monitor a particular SIP call for training purposes Hi, I've been asked if it is possible to allow a user to

I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the new box, I've installed a generic ebay X100P. I don't have my livevoip or voicepulse accounts set up yet on the new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP.

We are currently using Asterisk 1.0.8 and noticing a rather odd problem that occurs roughly during the middle of the day sometimes. It seems that Asterisk and Bellsouth are getting out of sync somehow on the the status of the PRI channels. I get warnings messages like: Jul 18 10:50:24 WARNING[744]: chan_zap.c:7585 pri_dchannel: Ring requested on channel 0/1 already in use on span 1.

Hi Someone complained in #asterisk-gui that the Zaptel hardware detection did not work. It seems that zapscan.bin did not work properly, as no /etc/asterisk/zapscan.conf was generated. Solution: detect zaptel hardware with genzaptelconf, and generate zapscan.conf from it. genzaptelconf grep '^fx[os][kl]s=' /etc/zaptel.conf \ | sed 's/\(fx[os]\)..=\(.*\)/[\2]\nport=\1\n/'