similar to: Announcement to caller when called party haspicked up - without initial Answer()?

This is super easy to do. All you need to do is to put that announcement in a MP3 and set a musiconhold class for that incoming Zap channel. Then basically when ever that PSTN number rings, Asterisk will play the MP3 stream "Your call may be monitored or recorded, please hangup if you do not agree...etc" in a loop until the line is answered. Caller doesn't pay a single dime to

Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So,

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

-----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, 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, I'd like to terminate IAX call on PRI interface. What conditions should be met to be able to send arbitrary caller numbers to called party, so he can call back to supplied ISDN number (that is different for every IAX user) - not through PRI interface, but plain ISDN call !! Thanks in advance, regards, Rob.

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 :

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in

Hi all, I have just been reading an article on the asterisk-doc site about ISDN X-Over cables. The article mentioned the converting of an NT1 to make this possible, has anybody got the information required to modify a BT NT1? Or any information on the BT NT1. Thanks in advance. Regards Dave

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have a general question about SIP access for nonregistered users. I would like to make it possible for basically anybody to make a SIP call to my asterisk without having to have a user account, but in a specific context. So that e.g. somebody could make a SIP call to SIP/stefan at my.asterix.pbx and it would go like this: [incoming_guest]