similar to: Callstatus on bridge IAX2 <-> ZAPTEL is always "answer" even if the call fails

Hi security@ list, In my self written, large ipfw rule set, I had something that passed http to allow me to browse most but not all remote sites. For years I assumed the few sites I had difficulty with were cases pppoed MTU != 1500, from not having installed tcpmssd on my 4.*-RELEASE, but then running 6.1-RELEASE I realised that wasn't the problem. http://www.web.de Still failed, &

I easily get confused when try to undertstand FXO & FXS ports. Is it possible to use an ATA to connect to a TDM400 card. If so, would I use FXO modules or FXS modules ? My goal is to connect my asterisk server to Vonage (via the ATA they send me) so I can use thier standard plan and do with out the Softphone account feature that only allows a few hundred minutes talk time. Thank you, Steve

Could someone tell me how the kernel used for kickstart (called the BOOT kernel in RH 9 days) is created? How does one generate this kernel using rpmbuild of the src rpm? I'm just trying to undertstand the differences. Thanks for any help.

Hi, Just wondering is there a way to make the width of bar charts that you create using R smaller? Also, a bar chart I created has a total number of 23 entries (with 2 different columns 14 and 9), how can I have the total number on the Y axis adding up to 23 and not the larger of the two columns? Hope this makes sense. BR Jack. -- View this message in context:

When I call my internal extension and hang up the phone keep ringing, I get: SIP/11-00000048 connected line has changed. Saving it until answer for IAX2/iaxy Is there a solution for it? SIP does not detect that IAX has hang up the line. -- Joseph

I am having a bit of a problem getting AMD to work on a new server. On my regular office server it works like a charm. I am running Asterisk 1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and I am using a SIP trunk to send out calls (the same one on both servers). Here is the output of a call on my office server: -- Attempting call on Local/0445540881644 at CC2 for

I've connected some analogic phone to some fxs modules on an analogic card. I want to disable by default the call waiting sound. I know that dialing *70 before to call the call waiting is disabled until the next call, but isn't there a setting or a dialplan command to set up this automatically? Thanks -- /*************/ nik600 https://sourceforge.net/projects/ccmanager

Dear FreeBsd gurus, I have a problem concerning users password and authentication policies. The goal is 1)make freebsd to lock users after 3 unsuccessful login attempts, 2)force users to change their passwords every 90 days I've done such changes in Linux distros, with various PAM modules.But in Freebsd it seems that i need to use login.conf file. Here I made necessary changes in that

Outgoing call on Zap channel always gives DIALSTATUS ANSWER even when BUSY? Has anyone run into this? Here is my conf files: Zaptel: span=823,1,0,d4,ami e&m=1-24 loadzone = us defaultzone=us Zapata: usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes

Hello. I'm running some asterisks in a small voip provider in Brazil and we're having some problems with a analogic channel bank. When I make calls using analogic extensions, I have a crystal clear quality, but the receptor have a lot of noise and echo. We tested the situation using SIP, E1 links, GSM links, analogic trunks (FXO ports) and we always have the same symptons My

Hi. I'm getting new lines for using with Asterisk. In my Telco they said I could choose between Analogic lines and RDSI lines... I've already bought a TDM400P with FXO modules. Can you give some hints on the differences between RDSI and normal Analogic lines? Would I have problems for using a RDSI line with the TDM? Any other issue in general? Thanks in advance, RODOLFO --- avast!

If i use AMD() or the code below, now the problem is the fax machine/modem detection and answer machine detection get detected as the same. If i need to seperate the two how do i do that? For example, if i use AMD() to detect an answer machine by saying any greeting exceeding 2.5 seconds is a machine, how do i distinguish between a fax/modem and a long greeting? ---- dave cantera

Hi, I've some analogic lines and I'm asked if it's possible to program an asterisk for "checking" the inbound calls without answering them, doing something like this: analog line 1 -----+---------- asterisk | \______ analog phone when a call enter, asterisk sense it and store its values (callerid, date and time, etc) somewhere, but

Hi all, I'm newbe with asterisk and i'm facing with this problem that i'm not able to solve. I've to put an asterisk box after an analogic pbx wich require a 0 digit to give the dialtone. So when a client ask asterisk to dial an extension it should 1) send the 0 digit 2) wait for the dialtone 3) dial the extension the client send. How can i obtain this result? Thank's in

Hi, I have been reading the acl documentation and it seems that a "connect acl" is not available. I need to limit the users that can login in an IP number, is that posible with dovecot 1.0? (i.e. only these users can login from the Internet) Or a new plugin should be written? It is complicated to do that? Thanks Oliver -- Oliver Schulze L. | http://tinymailto.com/oliver Asuncion

Message-ID: <479F2A63.2070408 at centos.org> On: Tue, 29 Jan 2008 07:30:11 -0600, Johnny Hughes <johnny at centos.org> Subject Was: [CentOS] Unknown rootkit causes compromised servers > > SOME of the script kiddies check higher ports for SSH *_BUT_* I only see > 4% of the brute force attempts to login on ports other than 22. > > I would say that dropping brute force

CentOS 5.2 Postfix 2.3.3 (Came Packed with CentOS) Dovecot 1.1.1 Dovecot-Sieve 1.1.5 Did a complete new fresh install. When I send a message to: test at wildpeacockstudios.com, I get two error messages as listed in the /var/log/maillog: (1) (lost connection with mail.tibonline.net[12.179.81.11] while receiving the initial server greeting) (2) status=bounced (local configuration error. Command

Hello everyone, Having recently gotten Broadvoice inbound DTMF to work (thanks Greg)...I am now running into a frustrating problem...when a call comes in to the BV number via a cell phone (tested with 3 different cell phones; albeit all on T-Mobile) the beginning of the IVR welcome audio is cut off. A call placed via a landline phone over the PSTN to the BV number does not exhibit the problem.

I've tried to figure out from the web site the minimum hardware cost to run a small office Asterix solution but I'm afraid I miss something: Let's say that I want to connect four/five analogic extension to the PBX. I have: - 1 computer as the server (with linux and Asterisk on it) - 1 dummy patch panel to connect all the analogic phones around the office What (and how many) cards

Hi, I am pretty new to Asterisk. I am trying to make sure some human being answers the phone not the voice mail machine. How can I programmatically identify that? Here is my Sub: sub DialPhysician { my ($self, $con, $PhysicianPhone, $call_id, $conv_id) = (@_); to_log($self, "Inside Dial Physician", 2); my $DocPhone = "1".