similar to: WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

hi there :) i get this error in the asterisk CLI: Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away when I run 'cdr mysql status' i get: Connected to voiceone at localhost, port 3306 using table cdr for 18 minutes, 24 seconds. Wrote 8 records since last restart and 1 records since last reconnect. I

Hi, I am having some problems mentioned below, the box is in production live environment with traffic around 30 - 100 calls. I am running T/E410P in a Dual P4 xeon with HT disabled. I am using zaptel 0.9.0 and asterisk stable 1 release. There is no gui, just mysql, perl (small script) and asterisk. System runs very smoothly if the calls are around 40-50 and comes one by one , however sometimes

Hi! anyone from here, who uses voiceone <http://www.voiceone.it> as their web gui for asterisk pbx? I know it's still under development but i wish someone would be joining on the development 'cause i think it's a great project to finish. I started some things on the validation forms on the zapata/zaptel part which is not included on the demo site. I hope I can get more help

Hi all! We've released VoiceOne 0.4.0, a web-based and open source solution which allows to fully manage an Asterisk service hosted on a LAMP server. We focused on an charming and overall user-friendly interface. Thanks to the authentication based on roles, once configured by a super user, the PBX may be easily maintained even by an Asterisk unskilled users. From a technical point of

I am using VoiceOne http://voiceone.it/ as my management interface. I am not 100% sure when it started, but my CDR is now full of "s" as the DST instead of the actual dialed number. As I understand it - it is because it is being recorded in the CDR while in a macro (as below). Is there any work around so that I can record the actual dialed number? [macro-dialout] exten =

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

Hi list! I?m connecting a Brasilian voip (- gvt.com.br -) provider through my asterisk box and using the register command from sip.conf. What I can?t understand is why my unit sends a new registration message every minute! And every time my asterisk box sends a registration, it gots a sucessful response, and shows de message: "Oct 14 16:48:22 NOTICE[4090]: chan_sip.c:8742

Guys, when registering I am getting this error message, my question is that if this could be the reason whay I am able to make calls but not to recieve call ? [Apr 3 11:24:31] WARNING[19578]: chan_sip.c:15104 handle_response_register: Got 423 Interval too brief for service +506phonenumber at domain.co.cr@host.ip.addr, minimum is 3600 seconds Thanks -- http://celord.blogspot.com/

Anyone come across this message: WARNING[2203]: chan_sip.c:9633 handle_response_register: Got 200 OK on REGISTER that isn't a register I get it from time to time but don't know what to do about it! Thanks, h

I've tried to install a couple of applications like the teleyapper script on nerdvittles but I seem to be getting an error executing the AGI script. I have also tried installing the voiceone gui and have problems executing the AGI scripts in this. Here is the error == Spawn extension (yapper-gen, s, 5) exited non-zero on 'IAX2/250-2' -- Executing

I've had a heck of a time getting a SIP channel to work in Asterisk 1.2.7.1 (Redhat 9.0). I've done it successfully a number of times on pre 1.2 versions of Asterisk so perhaps it's version related. Any insights are welcome! At first I wasn't able to dial out on the SIP channel _whenever_ I started Asterisk (i.e. not just when the box was booted). I always had to do a reload

Hello Asterisk People, I am having trouble connecting asterisk to a huawei SoftX3000 Switch, i can succesfully connect other softphones like Zoiper, but when it comes to Asterisk SIP Client, the system doesn't authenticate, i have the following configuration: peer: 10.220.0.2 username: 4857768 password: 4857768 the configuration is as follows: in the general section: register =>

What does that mean? Where can I get more info about it? Mar 18 07:19:51 WARNING[9563]: chan_sip.c:7549 handle_response: Got 200 OK on REGISTER that isn't a register

Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

Hi there, I want to use asterisk for sip comminication with max 1000 users Which gui shuld i use for adding users and managing asterisk? I tried AMPortal, it added extensions to mysql but asterisk did not find users i added ? installed asterisk 1.2.2 on FC4 Toygun

I installed a 2 port serial PCI card on a system running CentOS 4.3, but I can't get the additional serial ports to work. The card I used is from StarTech (http://www.startech.com/Product/ItemDetail.aspx? productid=PCI2S550&c=US). When it is detected by kudzu, it shows up as a "NetMos Technologies PCI 9835 Multi I/O Controller". I've included the output of