similar to: WaitExten not responding on key presses

Hi, I'm trying to store some encrypted values in hiera, using either hiera-eyaml or hiera-eyaml-gpg. While hiera-eyaml (PKCS7 encryption) works fine, the agent gives the following error message when using hiera-eyaml-gpg: Could not retrieve catalog from remote server: Error 400 on SERVER: Error from DataBinding 'hiera' while looking up 'the_secret_field': can't convert

Hi there, I'm getting a bunch of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors mean? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

I have verified this on two machines, but it would be helpful if others out there can reproduce it too. Also, I do not know if it is Xorg or the FreeBSD keyboard drivers, since I see no way to reproduce on the console (i.e. turn off repeat). In an xterm, type: "xset r off". Then try some multiple-key combinations (i.e. keep holding first key(s) when you type the next one): po (o does

The subject says it all. I'm betting there's a way to do it, but so far I haven't found the dialplan runestone via web searching. Thanks. b.

If '#' is pressed during a call the CDR that is written at the end of the call contains '#' in the dst / destination field rather than the number that was originally called. How do I avoid losing that original number so that I can use the CDR for billing? I've tried not having a '#' target in extensions.conf and I've tried calling ResetCDR(w) in the

In our IVR we have a user enter a 6 digit number and information if returned. Our problem is that no all of the digits that the user presses are being recevied correctly. Its not as if the first digits are being cut off or the last, its just some digits aren't coming through. Our setup is that we have an IAX channel coming in from VoicePulse going to our Asterisk box via the T1 coming in to

Hi We have a few conference numbers and all use MeetMe using the D option. We have noticed sometimes that the server is picking up more key presses than were actually done, i.e. the user presses 1234 for the pin and in the logs we see something like Created MeetMe conference 1022 for conference '12234' or Created MeetMe conference 1022 for conference '112334' Has anyone else

Hi, Does this report ring a bell here? I didn't check, but the image should carry an isohybridized 4.02 version of isolinux. The working (lenny) version is 3.71. Thanks, Feri. -------------------- Start of forwarded message -------------------- Subject: Bug#604245: Syslinux fails Date: Sun, 21 Nov 2010 14:13:41 +0100 From: Gyorgy Jeney <nog.lkml at gmail.com> To: submit at

Hello, I am using macros to give the ability to a call-receiver to 'accept' a call. However, any keypress connects the caller. Anyone have any suggestions about how to re-engineer this so that the receiver can deny the call, or press other keys to do other actions, without connecting to the user? Thanks, Jason Wolfe

Just released by the CPSC on their recalls mailing list; please forward to any venues where you feel operators or resellers of the SoundStation might be, with this preface included. My 2W had a battery with the part code 1520-07804-002; its date code was GP0806, and therefore predates the recall period. Cheers, -- jra ----- Begin forwarded message ----- This message consists of the following:

I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all

-----BEGIN PGP SIGNED MESSAGE----- CERT Advisory CA-2003-21 GNU Project FTP Server Compromise Original issue date: August 13, 2003 Last revised: -- Source: CERT/CC A complete revision history is at the end of this file. Overview The CERT/CC has received a report that the system housing the primary FTP servers for the GNU software project was compromised. I. Description

hello.... I'm new to wine and tested and installed it, because I wanted to run the hotline server from windows under linux... upload, news works great but download from this maschine is not working! strange thing does anybody has the hotline server runing? do you have had similar problems with another network app? this are the errors.... imted from C:\windows\system\wininet.dll, setting

Hi All, This is more of a hardware question that an Asterisk question so I hope this is still the correct place for the post. I know with the Linksys phones you can create a hotline by using the dial string of (S0<:number>). I have been trying to do this with a PolyCom phone but I have not been very successful. Does anyone know how to create a hotline phone with a PolyCom?

Hello, I have a problem with our asterisk server. (Version 1.4.15 RPM Build running on Fedora Core 7) After 3 day's running, i'v got this message in my logfile: Dec 14 13:14:09 amsec-tk1 asterisk[4321]: VERBOSE[11363]: -- Executing [9020 at AMS-UPPORT-HOTLINE:1] System("SIP/SN2400-67ec6aa0", "echo -e "Eingehender Support Anruf am 14.12.2007 um 13:14 durch die Nummer

Hi Samba-People, while Samba is doing really great as a server for windows clients, I'm currently not happy with way the program interacts with other servers, namely Netatalk or Mars netware emulator. It is no problem to open the same file on a Mac and a Windows PC read-write, thus inviting data corruption even without oplocks coming into the game. I understand that unix file locking modes

hello all i''m a newbie on mechanize, but already had some success. But now i''m in a real problem : The following HTML-Code is not parsed corect, not all form element are found. I''can''t change the server-code. How to fix ? HTML-Site ------X-----------------------------------------------------------------------X- <!DOCTYPE HTML PUBLIC "-//W3C//DTD

Anyone using the UIP200 with *? I am having difficulty getting the phone to register and *stay* registered for more than 4 seconds. The * console shows the UIP200 registering and records the user agent. The UIP200 displays station name and time on its LCD, then 4 seconds pass and it shows #1 DISCONNECTED. The * console reports no problem and believes the device to still be registered. A call

The following advisory was issued by CERT yesterday. Because it affects FreeBSD systems as well, we are forwarding it to the appropriate FreeBSD mailing lists. We would like to thanks CERT for cooperation with the FreeBSD security officer on this subject. -----BEGIN PGP SIGNED MESSAGE----- CERT Advisory CA-98-13-tcp-denial-of-service Original Issue Date: December 21, 1998 Last Revised

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>