similar to: Newbie: Mac OS X - Asterisk Gui 2.0 (svn) loops at " Verifying Dialplan Contexts needed for 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 >

Hi, I use asterisk branch 1.4 and gui 1.4 as well. I have the following situation: When I try to make gui configuration by http://localhost:8088/asterisk/static/config/setup/install.html I can see that application logs my user correctly but there is no browser window shift to the next page. it stays at the logging one. I get the following info in the console: [Feb 4 09:33:09] == Parsing

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

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

Hey Guys, We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? extension.conf exten => 7770,1,agi(allpage.agi) exten => 7770,2,meetme(7770,dq) exten => 7770,3,playback(beep) exten => 7770,4,hangup following is agi debug....

Hi, After upgrading a debian/lenny server to 1.4.26 I get this error: == Manager 'munin' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'munin' logged on from 127.0.0.1 [Jul 26 17:45:12] ERROR[12354]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe repeated each time munin logs in. Should I be concerned?

Hi gang, I have a "fun" one for you. I'm not getting the quality of sound I want out of GSM, so I'm trying to make my files into .WAV (.wav) format. Here is the "file" output for 5 files: file *.WAV cents.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft

Hello Incoming calls from the FXO trigger an AGI script which simply NOOP data sent by Asterisk through stdin. The first two NOOP work fine, but after this, Asterisk isn't happy: ============= extensions.conf ... [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/test.lua) exten => s,n,Wait(5) exten => s,n,Hangup =============

Hello List, I have been working on a little PHP software that uses AMI's UpdateConfig command in order to modify some of it's config files. I was working with 'Asterisk 1.4.22.1' and everything was working. After upgrading to 'Asterisk 1.4.23.1' I receive a lot of errors of the type: ERROR[11505]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe

Hi gang, Since I'm getting no joy from device_Status or SIPPEER in 1.4.26-rc1, I thought I would do an AGI to read my hints and check for line in use that way. The AGI works fine from a prompt, but returns the dreaded "utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I try to run it from the dialplan. Here is my dialplan snippet;

Hi All, Been having problem using the AMI, i've got this PHP script: $socket = fsockopen("1.2.3.4","5038", $errno, $errstr, $timeout); fputs($socket, "Action: Login\r\n"); fputs($socket, "UserName: amiadmin\r\n"); fputs($socket, "Secret: amiadminpassword\r\n\r\n"); fputs($socket, "Action: Command\r\n"); fputs($socket,

i have a problem with AGI in Asterisk 1.0.5, the problem occurs either with PHP or C AGI scripts/programs. Well, its simple, either asterisk is not sending correctly the command responses to the standard output, or for some unknown reason to me the scripts/programs are not able to read it from standard input. I have the next C test program for AGI: #include <stdio.h> main() { char

Dear, I am getting this in CLI on release candidate version of Asterisk. Any ideas, or points where to look? -- Launched AGI Script /var/lib/asterisk/agi-bin/rad-auth.agi [Oct 22 18:21:45] ERROR[9853]: utils.c:1126 ast_carefulwrite: write() returned error: Broken pipe -- <SIP/916-fc001968>AGI Script rad-auth.agi completed, returning 0 Best regards, Josip

I recently upgraded to asterisk 11 from 1.8. I had VXML working via AGI in 1.8 - from extensions.conf: [VXML] exten => s,1,Answer exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})}) exten => s,n,AGI(agi://localhost/url=${ENCODED}) exten => s,n,Hangup Using asterisk 11 on the same host with the same config in extensions.conf: -- Executing [s at VXML:1]

Hi List I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' most times it does what I expect and I get a response as below siptest:~# asterisk -rx 'dialplan reload' Dialplan reloaded. every now and then I get no response i.e. siptest:~# asterisk -rx 'dialplan reload' siptest:~# and a "verbose 10" setting shows [Mar

Hi, I'm trying to use StatEt IDE for Eclipse as my R editor, but I'm completely lost. I've read all I could find online, made apparently all I had to do (installing rj, configuraing StatEt, etc.) but still cannot make R running. Below is the error log file. Thank you so much for assistance. Matteo !ENTRY de.walware.statet.r.console.ui 1 0 2011-12-05 16:21:51.355 !MESSAGE

Hi All, I'm having problems getting results from a PHP file. This is what the CLI is showing. -- Executing [111 at internal:1] AGI("Console/dsp", "GoTalk.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/GoTalk.php [Dec 14 11:57:25] ERROR[20260]: utils.c:1019 ast_carefulwrite: write() returned error: Broken pipe If I run the PHP file from

ok it works i had a problem with a syntax: i had to wrire: exten =>_!X.,n(external),Dial(SIP/011212664800450 at pstn2,,S(20)) thanks ________________________________ De : Adil Zaaraoui <adilzeaaraoui at yahoo.fr> ? : Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Envoy? le : Jeu 8 juillet 2010, 19h 41min 15s Objet : Re :

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000