similar to: action at registering or de-registering

Hello For customers who need a small IP PBX to handle up to four ISDN lines (in France, so I guess that means EuroISDN) instead of a PC + Asterisk and an ISDN gateway box, has someone already played with the Atcom IP-4B? www.atcom.cn/IP-BRIM.html Any feedback appreciated.

Just curious, I noticed that with SetLanguage() you can change it into a lot of other languages. Yes one can record them easily enough with "record", but don't like to re-invent the wheel.. Browsed through a lot of google-pages but failed to find any other languages (except for FR and ES on the digium site) Any pointers for german, dutch, greek, italian, .... prompts? Hans --

Hi all, Perhaps someone can give me a hint i the right direction... Sometimes dtmf is recognized, sometimes not. I'm using 1.2.19 asterisk with misdn for my hfc card. When i got in incoming sip-call, dtmf is recognized, When i phone my self (isdn-phone or gsm-phone) no problem with dtmf When SOME (not all) people phone me (isdn-incoming) DTMF is not recognized. How come? Either it works

Morning! This may be the "day of the dead" in some regions, but we expect the usual lively discussion today at 9AM PDT, 11 Central, 12 Noon EDT, 4PM UK and Portugal, 5PM Paris, $deity-forsaken hour down under. This Sunday, I believe the USA falls back to Standard time. Future VUC are still at 12 Noon EST. Info site: http://voipusersconference.org PSTN (724) 444-7444 enter 22622# 1#

Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,

Hi, I've been looking at the list at: http://www.voip-info.org/wiki/view/chan_mobile But when googling of any of the "known working" devices, there ain't any for sale anymore, probably replaced by more recent types. So, anyone around here who bought recently an BT-dongle that is working with asterisk? hw

My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part

Hi List, i have to put an * between two other SIP gateways and due to some circumstances, i have to use sip over tcp. With 1.6.2.6 this is working fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B (ocs) and that's about it. In the other direction however (ocs -> me -> deverto4) the call setup is complete but there is no audio. I can see the audio in the form of

Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent "issues" that

Hello Everyone - Well, I think I'm getting closer with the asterisk connection. This is my setup and I keep getting this error below in ,my /var/log/asterisk/messages file. I have opened 5060 port on the firewall box. I would this is Warning which I can ignore! But I see the connetcion coming but NO DIAL TONE on mt ata186 box sitting in my 192.168.200.x site! I'm using ATA186(cisco

Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, could that also be multiple users? For instance, for each sip-user an xmpp-user? When i skim

I am not sure if anyone has done this but I am assuming it has been done before. Setting up a multihomed DHCP server accross multiple network blocks to issue PXE/BOOTP setup actions? Some background, I have successfully setup PXE to work with a cross over cable (first setup and testing env.), moved cient/server to switch and configured boot menu, images etc. and now after setting up the

I have tried to google for this kind of application, but have not found. There should be a web interface, where the client (person) could start an rsync backup of his/her workstation to a remote server (= the web server machine). I need this because the client has a dynamic ip, so the ssh key authentication on the server never works un-interactively - because every time the cient ip has

Hello, I'm trying to create a vpn using TINC with a WindowsServer2003 as client, but I'm receiving the following messaging when running "tincd -n vpn -D -d5" -------------------- ... Trying to connect to server <...> Timeout from server <...> during authentication Could not set up a meta connection to server ... -------------------- I tried the same tinc

Dear all, I'm trying to perform a principal components analysis of a sample of individuals, and to plot it in 3D, assigning different colors according to the population each individual belongs to. Given that the matrix I have to use for the PCA cannot contain cualitative variables (here, the population of origin), I have no idea how can I do this in R. ANy help? Thanks in advance

Dear list members, I am new to R and to the R-help list. I am trying to perform a mixed-model analysis using the lmer() function. I have a problem with the output anova table when using the anova() function on the lmer output object: I only get the numerator d.f., the sum of squares and the mean squares, but not the denominator d.f., F statistics and P values. Below is a sample output, following

Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? Regards Bilal

Ok, i tried the suggestion: Instead of: sippuser => resource, database_name, table_name sippeer => resource, database_name, table_name I put in: sippuser => resource, context, table_name sippeer => resource, context, table_name Unfortunately, with the same results. btw i tried both "general" as "default" Besids the commands i tried below, isn't there any

Hi all, Can anybody inform me wether the "RedHat Certificate System" or actually a CentOS equivalent is available for CentOS. Just skimmed on a download site through the RPM's for 5.3 and I couldn't find it. According to their pressrelease, it the code should be gpl, allthough I can't find any rpm for RH, FC or Centos. It seems that this is one of the few CA-packages for

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I