similar to: ISDN X-Over

Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk

For all the peoples that wanted to test my windows IAX2 phone, I've put it up on a server where it can be downloaded. For the ones that wanted the DLL, it's available on the same page. For the DLL, I will post a list of the functions in it and the parameters it expect as soon as I have some free time. All comments (good or bad) are welcome The phone can be used mostly with the keyboard

I've read quite a bit in the older mailing list posts and the wiki but I'm missing some simple point. 1) What is required to send an SMS to a mobile outside the office given: Channel: ZAP/1 send it to $SMS_RECIPIENT (which includes the final "extra" digit) via $SMS_CENTER=the national message center server for sending messages $MESSAGE= the message text How is the .call file

Hi, > I've never run against a commercial PBX that didn't need > maintenance. Acknowledged. > VM hard > drives fail, > ... > Asterisk is > every bit as stable > as the old-gen KSUs and PBXs. There are big differences. As I know of no other PBX that uses 'consumer' hardware, asterisk is also struggling with problems in the underlying Hardware. And

> Hi all, Hi Marco, > I was thinking about integrating an old PBX with Asterisk and I was wondering > some possible configurations. You didn't mention the number of lines your PBX uses, but think of a third scenario: Install an asterisk with twice the number of BRI/PRI-Ports your current PBX has. Connect half of them to your carrier, the other ones to your old PBX (Some sort of

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP

G'Day All, Can someone help me out please. My new CISCO 7960's manual says I have to setup a TFTP server. Googled it and got a little understanding, but from * standpoint, well I am still a lost. Can I set this tftp server on the same * box? Can in be on a WinXP box? Which tftp software would you recommend? Thanks much. BTY: Does anyone have a How-To on getting the 7960 fully configured

Is anyone successfully using this phone with Asterisk? There is a lot mentioned about CISCO but nothing about Nortel... Alex.

Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>

Hi! I am using OpenBSD 4.4 with the build-in TFTP-Daemon for PXE. Distributing OpenBSD works fine. Now I want to distribute OpenBSD and WinPE. I've installed PXELinux 3.72 and configured it. Works also fine for PXELinux and distributing OpenBSD. To distribute WinPE I need a remap rule (\ -> /) for the TFTP-Daemon. I've created a file /etc/tftpd.remap with the following rule: rgG \\ /

Hi, in case, anyone is interested... I have started compiling a blacklist of hosts and networks from which SIP fraud attempts occur. My criteria currently are: To block an IP: - Minimum 3 attacks within one week from the same IP To block a network: - Attacks from minimum 3 IPs from that network within 2 weeks Common criteria: - Provider does not react to complaints OR - Provider sends autoreply

Hi all, I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax

Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear busy Hangup Pick up again Dial a code like *123 => jumps into a context which redials until callresult is not busy Maybe like this: [autoredial] exten => s,1,Set(number=${CHANNEL(lastdialed)}) exten => s,2,Dial(SIP/${number}@account,60,g) exten => s,3,Wait(15) exten

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

You could try to use the native mp3 support for MOH if you really want mp3 support. It is a lot better than using mpg123 IMHO. mpg123 kept doing nasty things to my system :) See http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musicon hold.conf there is a section about the native support. Guillaume > -----Original Message----- > From: Stefan Gofferje

Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36

2015-01-09 3:53 GMT-06:00 Stefan Gofferje <lists at home.gofferje.net>: > > Do you really want to detect "ChallengeSent"? That should occur also on > legitimate login processes... > Hi , strange thing is that I still have not this asterisk in production and I see many attempts Connection. Now keep in mind that when a connection of authentication is successful the

On Tue, 15 Feb 2005 03:02:45 +0100, Stefan Gofferje <stefan@gofferje.homelinux.org> wrote: > Hi folks, > > I have some trouble with the FOP and would appreciate if anyone could > point me into the right direction. There is a FOP user list, although not too active. http://www.asternic.org/ > Is there a way to define a button like Zap/g1/6000 and have it light up > when

It's not just you. Same thing happens here. I went back to 1.0.7. Stefan Gofferje wrote: > Hi folks, > > I used to have some constructions like > > exten => number/callerid,1,Goto(somewhere) > > After updating to 1.0.8 those does not work any more. > Any hints? > > Regards, > Stefan >

Hi folks, when I start asterisk directly, I get a colored CLI. When connect to a already running asterisk with asterisk -R, it's never colored, despite I'm running both from the same console (tty). Is there a way to force asterisk -R into color mode? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security