similar to: Strange POP3 problem...

Hello, Seeing a strange thing with dovecot 1.0 b7 pop3. If a user gets a particular _very_ short spam message (not sure what virus makes these..), with NO Subject and NO To: headers and NO body, (example available upon request), the user is unable to download the mail. The log entry is: May 10 10:51:35 mail dovecot: POP3(user): Disconnected top=0/0, retr=1/1344, del=0/75, size=16383381

Hello everyone, ?? I'm new to TINC. Today, I've started using it, and now I've a network on around 9 dedicated servers and 15+ VPS ?? into a single VPN. I switched from OpenVPN. ? ? ?? But, I've a problem adding another node to the VPN. ? ? ?? Basically this node is a VPS with two IPs, on a single eth0, with eth0 -> IP1, eth0:1 -> IP2. IP1 and IP2 are on ?? different

Our law firm is looking for a contractor to assist in developing and extending our internal case management system. This system tracks most aspects of our workflow. Our current system is deployed as a WebObjects application and would need to be rewritten in Ruby on Rails. We need a dedicated person(s) to help us rewrite the system and drive future development. Applicants must have at

Hi, I''m trying to get RAdmin [uses tcp 4889] access to my Windows machine which is behind my firewall. I have zones: gbl : the world loc : my lan fw : firewall I placed the following in my rules file DNAT gbl loc:192.168.0.2 tcp 4889 - When trying to RAdmin I get a cannot conect to server error. 192.168.0.2 is my Windows Machines IP address. Can anyone help me? Shorewall 1.3.9b

I have a Fedora 5 server running Dovecot (currently 1.0 rc10). The spam filter (amavis) automatically tags email (using plussed addressing) such that messages flagged as spam are sorted into a "spambox" folder by procmail. Seemed like a good idea at the time, but because of some legacy issues with filesystem quota also being in place, combined with most users connecting with POP3

Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/25402ade/attachment.htm

Quoting asterisk-users-request@lists.digium.com: > > Would it be considered trolling to start a thread on Cleaning Maple > > Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi? > > Let's not forget the weekly "tooques and telephony" segment, and a > review of > the best block heaters for your wi-fi fones. > Oh, we're gonna have a good time next

Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. I have Allow as: ilbc gsm ulaw alaw -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667)

The PJSIP endpoint is configured for ulaw only. Not sure how or why we are seeing the g729 on calls for this endpoint. Would this be a case that asterisk detects the rtp stream is g729 even though it's negotiated as ulaw? Why would asterisk change the format to g729 when disallow = all and allow = ulaw are the endpoint settings? [121] type = endpoint context = IS transport = transport1 aors

Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with "no response to our critical packet". Calls to voicemail and internal extensions work fine. I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect

Anyone out there have a functional DUNDi configuration using SIP for the inter-Asterisk transport? I've gotten it to work with IAX2, but if I change it to SIP it does not pass the call over even though it knows where to send it. Thanks. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally

Hi All, I have the data like this : >sample <- read.csv(file="sample.csv",sep=",",header=TRUE) > sample stdate Domain sex age Login 1 01/11/09 xxx FeMale 25 2 2 01/11/09 xxx FeMale 35 4 3 01/11/09 xxx Male 18 30 4 01/11/09 xxx Male 31 3 5 02/11/09 xxx Male 32 11 6 02/11/09 xxx Male 31 1 7 02/11/09

Hello, First of all sorry for my English. I am experiencing with Samba and I have a problem. I have an old server (OLD) with Red Hat 9 and Samba 2.2.7a that is working well. Now I try to start up a new server (NEW) with Red Hat Enterprise 4 and Samba 3.0.22. If I try to connect from NEW to itself by using smbclient I got the shared resources list correctly. If I try to connect to NEW from OLD,

---------------------------------------------------------------------------- ---- FULL RSYNC LOG ---------------------------------------------------------------------------- ---- /usr/bin/rsync -av --force --delete-excluded --exclude-from=/usr/local/etc/snapback/snapback.exclude -e ssh --delete peru.cbm.mercyships.org:/ /home/backup/peru/hourly.0/ <bunch of lines deleted> wrote 873039

Hi, I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX. Calls in and out work fine, as does voicemail. The PBX at the Data Centre has an External IP, Nat?d to it by the firewall, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this

Hi, I'm currently setting up an Apache SSL reverse proxy for Exchange 2003 Outlook Web Access. The setup that I have works fine on my Gentoo laptop or on a Trustix server, however, when I try to set it up on an RHEL based distro, with the exact same virtual host settings, I get some weird error with the authentication mechanism. I have tried with both CentOS 4.2, based off the server CD

Dear all, I?m testing a server and try to simulate a server in production. We have a SSL certificate and I have configured the test server with the same servername as it is in production. To access it, I change the hosts file in my laptop to reach the test server. However, the Java application running in the server tries to access some local web content. I have changed the hosts file

Hi List, we're calling a sip phone from our Asterisk Server, and try to add logic depending on the dialstatus Stripped down example; exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =

Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI> show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2

Hi, I have been trying to get the cph() function of the Design package to work but get an error message I don't understand: Error in if (!length(fname) || !any(fname == zname)) { : missing value where TRUE/FALSE needed I have tried the same for a dummy dataset I made, and than it seems to work fine. However, it doesn't do it for my own data. Does anyone have a clue as to what