similar to: VoIPGMap: Graphing active Asterisk calls on Google Maps

OMG.. how embarassing.. that was my personal reminder E-Mail for x-mas dinner. Not meant for this list. Please ignore. Shame on me.. *blushing* LOL. Am 12.12.2014 um 21:19 schrieb Markus: > Anna Crepes: Traubenzucker > + Feldsalat spezielles Dressing (bringt selbst mit?) > > > > -------- Weitergeleitete Nachricht -------- > Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26.

Anna Crepes: Traubenzucker + Feldsalat spezielles Dressing (bringt selbst mit?) -------- Weitergeleitete Nachricht -------- Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Thu, 11 Dec 2014 15:34:39 +0100 Von: Markus <universe at truemetal.org> An: universe at truemetal.org Geschenke Moritz: dunkle Schokolade. Geschenke Anna: normale Schokolade. -------- Weitergeleitete

Hi list, n00b question, but I can't figure it out: [callthrough] exten => _+X.,1,NoOp(nothing here) #include "blockedall.conf" exten => _+X.,n(hangup),Hangup exten => _+X.,n(nohangup),GotoIf($["${CALLERID(num)}" = "anonymous"]?nocli:cli) ... more stuff that is handling the call ... I'm putting CLIs that I don't want to be able to call my

Dear list, on Asterisk 1.4.21 which is being used in a callthrough scenario - callers call via PSTN to a DID coming in via SIP and then dialing outbound via DTMF and the outbound calls get routed via some SIP termination provider - lately I see that every now and then MusicOnHold gets triggered like this on outbound calls: Started music on hold, class 'default', on

Markus <universe at truemetal.org> schrieb: > I don't think so. Most users will use the router provided by Telekom. These users do NOT use Asterisk on theis Telekom-line... I asked for someone using Asterisk on Magenta Zuhause... :) > Anyway, after 15 seconds of Google'ing for Magenta Zuhause and SIP, > maybe this will help you: I already know these links, and I

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

On 14/01/2019 07:09, Jobst Schmalenbach wrote: > Hi > > Specs in subject line: CentOS 6.X all latest patches), iptables 1.47, Apache2.2 > > I use the Geolite legacy databases together with iptables 1.47 to filter traffic for a variety of ports and only allow .AU traffic to have access. > I use ipdeny's aggregated country lists to do the same thing:

Hi Specs in subject line: CentOS 6.X all latest patches), iptables 1.47, Apache2.2 I use the Geolite legacy databases together with iptables 1.47 to filter traffic for a variety of ports and only allow .AU traffic to have access. Maxmind (https://dev.maxmind.com/geoip/geoip2/geolite2/) changed the default DB to the latest version which is GeoLite2, this leaves all users in need of the old

I have an asterisk with a TE110P configured as T1 which is behind a PSTN gateway. This gateway has an E1 to PSTN and a T1 to asterisk. This T1 is configured as Network and * as CPE. Every call I receive in E1 gateway is directly switched to asterisk using T1. Remember E1 is alaw. Both E1 and T1 have Natural Microsystems boards with a very simple software. When I call to E1 asterisk signalling

First time subscriber/poster My attached patch allows GeoIP support for the free GeoIP database and C api from http://www.maxmind.com/ Install GeoIP before compiling and it should work. It adds 2 options to sshd_config DenyCountry Deny access from a specific country based on GeoIP lookup. Use multiple DenyCountry entries to deny access from multiple countries. DenyCountry takes precedence

> -----Original Message----- > From: centos-bounces at centos.org [mailto:centos-bounces at centos.org] On > Behalf Of TE Dukes > Sent: Sunday, August 28, 2016 10:36 AM > My home system on a DSL line is getting worn out by bad behavior robots. > Awhile back, I created a .htaccess file that block countries by IP blocks. > Its 2MB in size. ... > So, today, I tried

On Tue, Jan 15, 2019 at 07:43:02AM +0000, Phil Perry (pperry at elrepo.org) wrote: > On 15/01/2019 01:29, Jobst Schmalenbach wrote: > > On Mon, Jan 14, 2019 at 07:29:45AM +0000, Phil Perry (pperry at elrepo.org) wrote: > > > On 14/01/2019 07:09, Jobst Schmalenbach wrote: > Below is my script for creating/updating an ipset to block my top 10 > Hope that helps Thanks, it

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

hi, today i upgraded from asterisk 1.4.21 to 1.6.2.9 (i know this release is not supported anymore, please don't tell me to upgrade). unfortunately now i can't use the rxfax() application anymore. i tried to use ReceiveFAX() the way i used to use rxfax() (multiple dedicated fax extensions getting their faxes from isdn via zaptel/dahdi), but this doesn't work. either fax

Hi list! I'm trying to make a SIP test call from Bria and/or 3CXPhone from a PC behind NAT. From Bria/3CXPhone I connect to an Asterisk 11.25.0 server on the internet at 100.100.94.210 with a SIP account "3333" created in sip.conf: [3333] type=friend secret=something host=dynamic nat=yes qualify=no disallow=all allow=alaw allow=ulaw canreinvite=no context=voipin I dial +1234

Hi list, I have a fresh install of Asterisk 12.0.0 and I'm going to use it only as a client. I'm trying to SIP REGISTER with a remote SIP provider. The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of the VM performs static NAT from the RFC IP address to a dedicated public IP address, however, they are rewriting ports at will.

Hi list! I'm getting this error frequently: ERROR[25193][C-0004f387]: cdr_mysql.c:203 mysql_log: Cannot connect to database server localhost: (2026) SSL connection error: SSL_CTX_set_default_verify_paths failed Right now, as a workaround, I reload Asterisk via cron once an hour, and after the reload everything is fine again _for a while_. Still, over the course of a month I lose about

Hi list! I think there are many german users in this ML, that use Asterisk with the new line of Deutsche Telekom (Magenta Zuhause). My ISDN will be converted in Juli (Kaboom-day at Juli, the 3rd...) and right now I can just hope, that I configured my Asterisk well to work with Deutsche Telekom, but I cannot be sure, since I can't test it... So my question: can someone using Asterisk with

Hi, Problem solved. It was related to SELINUX. The moment it is disabled, BIND service started properly. -- Thanks & Regards, Anantha Raghava DISCLAIMER: This e-mail communication and any attachments may be privileged and confidential to eXza Technology Consulting & Services, and are intended only for the use of the recipients named above If you are not the addressee you may

Hi, Firstly, what I'm trying to do is: * Get asterisk to pick up a SIP call via a DID * Prompt the user * When the user puts in a number, go to IAX.conf and route it according to what I've specified there, i.e Least Cost Routing, etc. I've set-up something similar to what I've found online, but it doesn't work! Asterisk doesn't pick up the call at all..... :( The files