search for: firedrak

Just to bring everyone up to date, could we get a report on the status of support for the 2.x protocol? The home page says "next major release" - is that 1.3 or 2.0? And is there any feel for when it'll be generally available? -- John Hardin Internal Systems Administrator Apropos Retail Management Systems, Inc. <johnh at aproposretail.com>

...a mobile phone (UTStarCom GF-210) that uses SIP MESSAGE to send "SMS" messages over VoIP. My Asterisk 1.4 installation drops these messages and returns a failure condition to the phone: [Feb 9 10:17:22] WARNING[11960]: chan_sip.c:9859 receive_message: Received message to <sip:2542 at firedrake.org> from "Display Name" <sip:mob776 at firedrake.org>;tag=87739132, dropped it... Content-Type:text/plain; charset=UTF-8 Message: test message (Packet trace shows a SIP MESSAGE, answered by a 405.) ...and apparently is unable to originate them either; SendText, which look...

...pe is to have a stable version released before Jan 1. At this point the main holdup is Solaris. I have had to disbale direct downloads from violet.ibs.com.au, demand for OpenSSH is saturating our little ISDN connection. I notice that: ftp://ftp.localhost.ca/pub/openssh/files/ (Canada) ftp://ftp.firedrake.org/openssh/files/ (UK) ftp://thermo.stat.ncsu.edu/pub/openssh/files/ (USA *only*) have already updated. Regards, Damien Changelog: 19991216 - Makefile changes for Solaris from Peter Kocks <peter.kocks at baygate.com> - Minor updates to docs - Merged OpenBSD CVS changes: - [aut...

I am working with a simple "follow-me"-style service: rather than have something that rings several phones in turn, the user dials a number (in the present implementation, unique to that user) to register his presence at a particular extension. What's the standard way to protect this from unauthorised use? Voicemail()-style, where the user has to enter a PIN once the connection is

The desired result is that user A's phone rings; when he picks it up, user B is dialled, and user A's channel is connected to that. (This is to be a back-end for a web-based address book.)

Hi, I'm trying to setup Asterisk so that: 1. I call a specific number that goes to a defined extension from my phone (an external line). 2. Asterisk notes my phone number (the CLID) and hangs up without picking up the call. 3. Asterisk initiates a call to my phone and prompts me for a passkey. 4. Asterisk validates the passkey and lets me enter another number (say FOO). 5. Asterisk dials FOO

I now have an OpenVox A400P and it is working well. Thanks to Ade Vickers for the recommendation, which I second. However, I need to make a slow transition between a conventional multiple-extension setup and a full VoIP network on these premises. So at the moment the Asterisk box shares the PSTN connection with several conventional analogue handsets. The desired result for an incoming call is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 1.2.1pre25 is out. Please use a mirror: http://violet.ibs.com.au/openssh/files/MIRRORS.html The following mirrors already have it: ftp://ftp.localhost.ca/pub/openssh/files/ ftp://thermo.stat.ncsu.edu/pub/openssh/files/ http://www.firedrake.org/openssh/files/ Changes: - - "Corrupted check bytes on input" when using triple DES has been fixed - - Added support for directory based lastlogs. This should make Irix as functional as the other platforms. - - Compilation fixes - - Documentation updates - - ssh-agent now...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 1.2.1pre25 is out. Please use a mirror: http://violet.ibs.com.au/openssh/files/MIRRORS.html The following mirrors already have it: ftp://ftp.localhost.ca/pub/openssh/files/ ftp://thermo.stat.ncsu.edu/pub/openssh/files/ http://www.firedrake.org/openssh/files/ Changes: - - "Corrupted check bytes on input" when using triple DES has been fixed - - Added support for directory based lastlogs. This should make Irix as functional as the other platforms. - - Compilation fixes - - Documentation updates - - ssh-agent now...

Hello, I'm trying to compile DAHDI on DEBIAN but i have the following error: root at Sangoma-Testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 What should i do? Thanks! -------------- next

Hi ! I have a python script that create and move .call files to /var/spool/asterisk/outgoing Sometimes...(in this case after 500 successfull calls) Asterisk don?t make the calls and the .call files are in the "outgoing" forever... Any Ideas? I'm using Asterisk 1.4.22 (in 1.4.36 was the same behavior) In my python script I move .call files using ... import shutil

I have a pair of Asterisk servers which are happily routeing VoIP calls. I want to hook one of them to the PSTN. Given that I am in the UK, what is a reasonably easily-available device to provide an FXO interface from a Linux box, with a minimum of faffing around with drivers? Just one line is needed, though in theory two might eventually be useful. My usual white-box hardware suppliers don't

The relevant part of my setup is something like: SIP phones -> local server -> remote server -> SIP-to-PSTN provider I want _some_ of the SIP phones on the local server to be able to get access to SIP-to-PSTN, but not all of them. The local-to-remote connection is IAX2 over VPN. Do I need to set up two separate IAX2 connections, one "privileged" and the other not, or can I