similar to: Call Forward on SIP unreachable (network failure)

Hi, I looked for asterisk in /usr/sbin using the commands ls and find and whereis and it was not there. I know that the process is killed because when I start asterisk using the command asterisk -vvvvc it starts and then it exits and the word killed is wrote on the console. Ever time I copy a new executable to /usr/sbin either using cp command or make install it gets deleted too. Now I used

Hi Thank you for your support. The server is actually compromised, I discovered that after making a deep trace using the audit daemon and looking for the kill signal (SIGKILL) that terminates asterisk. I discovered that there is an executable with a random name in the /boot folder that is killing and deleting asterisk !!! This executable is launched by a service in /etc/rc.d/ with the same

Hi, I am struggling with ?a very strange issue I have been facing for the past week;I have a fresh install of CENTOS 5.11 and I have installed asterisk 1.8.32 form sources.The asterisk installation went fine but as soon as I start asterisk executable it loads everything and then after the "Ready" line the process gets killed and when I try to run it again i get: /usr/sbin/asterisk :

So far these people let me know there are going to be there, who else is going and wants to do some networking ==== Joachim Vanheuverzwijn (zoachien AT securax.org) - Attractel.com - wednesday / thursday. Tan Aksoy - Telappliant - wednesday / thursday Antoine Megalla - SAND - wednesday / thursday loic didelot - wednesday / thursday Olle E. Johansson - edvina - wednesday / thursday Marius

Hi, I have been trying to get T.38 to work with clients behind NAT for the past week but with no success. I have an asterisk server on the public internet and several Grandstream (I tried Linksys too) HT502 ATAs behind NAT in different locations. I tried every possible combination of NAT, canreinvite, t38pt_usertpsource entries, I even tried asterisk 1.4.19, 1.4.24.1, 1.4.25 all with the same

Hi, Did anyone try to install Asterisk on the Mini-Box M300 with a Versa mini-ITX board 1GHz VIA x86 CPU? The box looks promissing, but I am not sure if Digium cards are compatible with the mother board (Versa mini-ITX) Also I am not sure if the 1GHz VIA processor can handle a Digium 24 port analog board, or an E1 digital board. If anyone had tried the Mini-Box, the processor, of the mother

Hi, As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Did you take a look at /var/log/syslog? Am 26.11.2014 21:08, schrieb Antoine Megalla: > Hi, > > I looked for asterisk in /usr/sbin using the commands ls and find and > whereis and it was not there. > > I know that the process is killed because when I start asterisk using > the command asterisk -vvvvc it starts and then it exits and the word > killed is wrote on the

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

Am 26.11.2014 11:37, schrieb Antoine Megalla: > Hi, > > I am struggling with a very strange issue I have been facing for the > past week; > I have a fresh install of CENTOS 5.11 and I have installed asterisk > 1.8.32 form sources. > The asterisk installation went fine but as soon as I start asterisk > executable it loads everything and then after the "Ready" line

After installing a new box and asterisk. i have got these errors [root at localhost ~]# asterisk Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory [root at localhost ~]# asterisk -vr Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) I didn't find a folder called asterisk in the directory /var/run [root at

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

Well, I had exactly the same issue as you described. It turned out to be a piece of malicious software that was running on the server. The customer server was compromised due to a weak root password and only Asterisk process was the target of the malicious program that was embedded deep into the server. The exact details escape me, but I do remember that it took more than two days of tracing

Sds, Paulo Henrique Cardoso Administrador de Redes - T.I. NHS Sistemas Eletr?nicos Ltda Av. Juscelino Kubitschek de Oliveira, 5270 Cidade Industrial, Curitiba - PR Fone/Fax: (41) 2141-9246/9247 www.nhs.com.br IMPORTANTE: Esta mensagem, incluindo quaisquer anexos, ? endere?ada exclusivamente ao seu destinat?rio e poder? conter informa??es confidenciais. A revis?o, distribui??o,

I gueess it is not a DNS problem, because either using name or IP, I have always the problem. I guess the problem is that I am using ssh on inetd.conf (sshd -i), so It has to generate a key each time I start a session. What do you think ? -----Mensagem original----- De: Dan Kaminsky [mailto:dan at doxpara.com] Enviada em: segunda-feira, 10 de junho de 2002 20:51 Para: Jorge Cleber Teixeira de

Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an inbound route! It matches a DID number. How can I route an INVITE sip:s at myip.com? The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it

How can one transform a character string into an argument of a function (which is not or I don't want it to be a character string)? Example: > expand.grid(c(1,0),c(1,0)) ## OK Var1 Var2 1 1 1 2 0 1 3 1 0 4 0 0 > paste(rep("c(0,1)",2),collapse=',') ## to be used below [1] "c(0,1),c(0,1)" > ## string is the input I want, but it

Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any