search for: girlfriends

Hi, Some background: I would like to call my girlfriend over the internet. We are both behind a nat router and I want to avoid portmapping. I've heard that you can call someone behind a firewall (nat router) with the IAX protocol, but I'm not sure. The questions: Do I have to set up my own PBX asterisk server or are there any other (free) servers where I can register on and connect

Ex-girlfriend logic not working in latest CVS? Incoming sip calls don't work. Anyone else seen this problem? Extension logic looks good: exten => 6153248305/_931NXXXXXXX,1,Queue(queue1); exten => 6153248305/_615NXXXXXXX,1,Queue(queue2); ;exten => 6153248305,1,Queue(queue3); show dialplan looks good: -- Added extension '6153248305' priority 1 (CID match

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

[This email is either empty or too large to be displayed at this time]

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for

I have this [TRONCAL-SIP] exten=>225/91,1,Answer exten=>225/91,2,Echo exten=>225/91,3,Hangup exten=>225/92,1,Answer exten=>225/92,2,Playback(conf-invalid) exten=>225/92,3,Hangup When I make a call CLI> -- Recv IAM CIC=8 ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=>225,1,Answer Works ok -------------- next part --------------

Hey all, Someone's posted one of my 800#'s on a poster in California for free concert tickets, so I'm getting calls from California AC's at all times of the day asking for tickets. I'm just using the 800# for friends and family, and don't know anyone in these area codes, so I'd like to just give these callers either congestion or a prerecorded message. Works fine

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

I decided to start again and delete all windows applications. I followed the direction on the Wiki FAQ to type Code: rm -rf $HOME/ .wine into the terminal and now everything is gone... Just when was thinking how happy I am with the way my laptop was set up all my files pictures backgrounds etc are gone. All of my personalisation of Ubuntu went along with my files downloads... i take it

it's funny. I only started this so my ex-girlfriend on the other side of the world could hear a song (downloading mp3's was not possible for complex reasons but streaming audio was) but goddammit i'm going to make this thing work if it kills me. At 07:29 PM 4/5/2001 -0700, Jack Moffitt wrote: >On Thu, Apr 05, 2001 at 06:21:37PM -0700, William Goldsmith wrote: >> > why

I am using Debian, and I am trying to write a program for my girlfriend to use in windows. And people suggested Visual C++, but when I will installing it with wine, it says that c run time error, does anyone know how to fix that -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.winehq.org/pipermail/wine-users/attachments/20070107/6a11cb7f/attachment.htm

My girlfriend is currently running Leopard and she wants to use a certain playstation game playing program. I set her up with a trial version of Crossover Office but it still has a few undesirable bugs. I'm an experienced Linux user with lots of knowledge of Wine's configuration options but not enough experience with Macs. Options?

Good Day To You All, I have installed Ubuntu 9.04 on a Laptop running Vista Business. When I try to start my little machine, GRUB keeps telling me there is an error: Either: Error 25, Error 21, or Error 5. Once in a while, it is 'Error 22'. And then there is no way to go any further: End of the line. The 'Error' crashes me into being nothing. Why do I get these 'Error

Hi, I am a newbie to Asterisk; need help understanding three-way conferencing & call-transfer features implemented over standard extensions i.e. on a TDM800P card (4 FXO + 4FXS) In Asterisk I have observed that if an extension is already participating in an active call (e.g. Ext A & Ext B communicating): 1. An incoming call to one of these active extensions would be presented with

We have asterisk running on a quad processor dell. The kernel has been compiled with SMP. However, asterisk seems to only use 1 processor. 3 of the 4 always stay at 100% idle. Is it pointless to have a multi-proc machine? I was going to buy a new dual 3.6Ghz Xeon server but if nothing will take advantage of the other proc... Perhaps my conception of multi-proc/threaded is warped. If asterisk is

Hi, please excuse my poor englisch. Is it possible to connect a (privat Test-Asterisk) to my privat ISDN and allow him to only answer one dialed number? We have 3 up to 10 Numbers on each (Euro-)ISDN (2 b-chanels), it cant't be done by the last Digits cause the numbers are completely different. For Example: I have 3 Numbers (641717, 928752....) Is it possible to tell Asterisk (in

I made the observation that I'm able to make a call with my SIP client (kphone) even when I'm not registered/authenticated. Of course, when I'm not registered at asterisk, people can't call me, but it's still a huge security hole, that unregistered Clients can make calls. Is there a way to tell asterisk to only allow registered clients making calls? I know about the

I have encountered a problem so I am posting this question to both the technical list as well as the normal user list for a solution. When I initially joined the samba box to the Win2k domain as a member server I followed the Samba 3 HOWTO and specified the OU in the Active Directory. After speaking with our boss this is not the correct OU for the users we need to authenticate. My question

Interesting read in Alec Saunders blog today. http://saunderslog.com/2007/03/01/mashable-telcos/ Thought it may interest some people on this list. As food for thought, why it is that ITSP's haven't come up with more 'interesting' voice applications? When asterisk first became available I thought it was the beginning of seeing really neat applications, think Verzion's