similar to: Phishing attempt posing as digium

If you are having trouble reading this email, read the online version<http://now.eloqua.com/es.asp?s=491&e=78675&elq=55426a8b6c714f5bb6f2bf4b5d37bf55> . <http://app.en25.com/e/er.aspx?s=491&lid=215&elq=55426a8b6c714f5bb6f2bf4b5d37bf55> Dear Lito, *The information in this email is given to you in advance to make you aware of an impending product release

Is the "Whats New at Digium the Asterisk Company" message I got from digium at en25.com really from Digium? If so I suggest to send it from digium.com and not to use those shady Eloqua redirect URLs. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk?

Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/<number>@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc).

Hi Sorry to use this public place, but IRC and emails to webmaster at digium have not helped in the past. I have several issues with using the files server downloads.digium.com, which has replaced the simple ftp/http file server ftp.digium.com. In downloads.d.c the directory listing is served through a seperate per-directory script with an obscure name. Let's look at

FYI/Heads up, I /just/ received what looks like a phishing attempt for information about Open Source PBX usage. It says it comes from Digium but all the links (including the one for digium.com) point elsewhere. Rod --

Hi, Are those messages from digium at en25.com messages from Digium sent via some kind of spamming service? I did not subscribe to anything at en25.com so why would I have to unsubscribe? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? ->

I know I must be being daft, but is there a way to set which context the queuing system uses when it dials the operators/agents? By default it appears to use the default context. I've looked through voip-info.org and can't find anything, someone please put me out of my misery.

Does anyone know why when calling out with an ATCOM AG-188 registered with IAX (haven't tried SIP), there is no ring. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070222/a4f29a97/attachment.htm

I am compiling svn head LLVM at the moment and I keep getting requests for password, which never used to occur before, followed by: Connection closed by 147.32.84.4, which is some site in Amsterdam. This occurs at this point in the make execution: make[3]: Entering directory `......./llvm/lib/Target/X86' Is this genuine, or has someone hacked the code and inserted some phishing attack into

Yes it is true and the very reason why i want all my Google Group Friends to be aware of the latest scams. They are not listed on the Scam Buster sites simple because they go unnoticed , well people like me get duped though. Please read <http://workfromhomedepot.blogspot.com/2008/02/latest-email-scam-fraud-phishing-online.html>and forward to all the people you care for. -- Yours Sincerely

This appears to be not as bad as I first feared. I had been using llvm previously on a remote machine and for some inexplicable reason a fresh compilation on my local machine wanted to make a connection to the remote one, in the context of: Updated Intrinsics.gen because Intrinsics.gen.tmp changed significantly. Though why this should be necessary, or useful, is beyond me.

Sorry about taking up the bandwith here, I have solved this interesting bug of my own now. It turns out I had inadvertently inserted a shell script into my search path which happened to have the same name as one of the more obscure system programs used in the LLVM Makefile execution and this was trying to connect to the remote machine instead of getting on with the job.... a warning against using

[ Sorry, I did not see the renamed thread until I'd already replied on the old one. Calling this a phishing attack is exactly right. ] On 2010-01-30, Joachim Schipper wrote: > If I understand you correctly, you argue that connecting to malicious > hosts is currently secure, and will remain secure, but that it will > become easier to convince people to send the passphrase for

I am inquiring about setup of the SWAT utility I have installed Red Hat 7.0 with samba installed during the initial setup of Redhat. I have two network cards installed in my Server and I am connected to the internet via a Cable Modem. When I try to start SWAT netscape displays the message that it cannot find the local host on port 90. I have downloaded the book over samba and I have also tried to

I have an asterisk 1.2.9.1 machine with zaptel 1.2.6 running. On the TDM400P, I have 1 FXS port and 3 FXO ports. dmesg reveals: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.6 Echo Canceller: KB1 PCI: Found IRQ 10 for device 01:01.0 PCI: Sharing IRQ 10 with 01:05.0 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

I have been seeing this problem for a long time and it occurs in 1.4.0b2 (as well as 1.2.0-1.2.12.1). If the internet connection is lost and I have SIP services that require me to register, any SIP devices attached to the system stop working. I have an IAX phone connected to one of my servers that I've been having this problem with which will work fine (and filover to the PSTN) the

I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a run, I have to admit. Asterisk itself only segfaulted once or twice, but the dns issues have been bothering me. And the box just needs to go. Everything is going on a Ubuntu 6.06TLS server, that's been perfectly stable. I had 1.4.1 installed and running, but not configured. Yesterday I upgraded to 1.4.11,

The Asterisk and Zaptel development teams have released Asterisk 1.4.1. This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability. It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety

The Asterisk and Zaptel development teams have released Asterisk 1.4.1. This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability. It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety

Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) Which should allow me to listen to conversations that hit the first (Set