cyber-cottage.co.ukI came up with this idea while on holiday many years ago and wanting to watch catchup tv on my ipad. But with everyone now wanting to make video calls thought would revive it
Its a simple cardboard stand that lets the tablet stand up.
The best cardboard to use is the “twin” walled type with a corrugated core. You can adjust the size so can be used with tablets or when made smaller with phones.
Make sure the folds are in line with the corrugations as this adds the strength
Hope you enjoy it.
We have various customers that have firewalls that only allow known trusted IP addresses through. Normally our office and our monitoring platform for example.
But if we are out and about we still sometimes need to access a system and its GUI, so we have created the simple script below that makes a ssh connection to the customer server and also tunnel to access any web gui.
This script is in place on the monitoring server so we can just ssh in to the monitoring platform and run the script. all that is needed is a single tunnel setup on the ssh client that i’m accessing the monitoring platform from.
#!/bin/bash echo ssh tunnel tool. 2013 cyber-cottage.co.uk echo Setting up a tunnel to $1 whois $1 |grep netname if [ "$1" = '' ]; then echo "You have no remote destination set" echo "usage: remotetunnel.sh <remote server> <remote ssh port> <remote system port>" echo "For example remotetunnel.sh 81.22.23.24 8022 80" exit fi if [ "$3" = '' ]; then echo "usage: remotetunnel.sh <remote server> <remote ssh port> <remote system port>" echo "For example remotetunnel.sh 81.22.23.24 8022 80" if [ "$2" = '' ]; then echo "You have no remote ssh or system port set, Setting ssh to port 22" port="22" else port="$2" fi echo "You have no remote system port set, Setting remote to port 80" rport="80" else rport="$3" fi if [ "$port" = '' ]; then port=$2 fi echo Remote system IP is $1 echo Remote ssh port is $port echo Remote system port is $rport read -p "Is this correct? (y/n) " RESP if [ "$RESP" = "y" ]; then echo "Glad to hear it" else exit fi ssh -L 9999:localhost:$rport $1 -oport=$port
In the past we have found that IAX@ peers have been reliable and solid.
But lately with the advent of bonded ADSL lines and other forms of aggregated lines we have seen issues where the IAX2 trunk will go down and a simple reload of Asterisk or even a restart doesn’t fix it.
Taken from Voip-info
This is something I’ve run into myself and my VOIP IAX2 provider has this issue with many clients running Asterisk on TrixBox or other custom made systems behing a NAT (Linux) router.
If our PPPoE goes down, we have to reboot our Asterisk server to get our IAX2 trunk to re-register otherwise, it will try and just keep timing out. I have the 4569 forwarded internal (Pierre Belanger adds: in many cases, the 4569 port forwarding useless unless your Asterisk server provides service to IAX2 phones from the Internet, i.e. not on your local LAN).
I have a dirty script that avoids having to reboot the TrixBox and restore our service within 2 minutes of a blip automatically, and logs the ‘blips’ so i can see how ‘reliable’ our service is.
We have take the original script posted and made some changes, Notably it checks a defined peer name as we have seen that the problem doesn’t always affect all peers on a system.
======Code follows ======
#!/bin/sh #We record the status of the IAX2 Trunk cd /root/ # I have script live in root, # Set the peer name to monitor here # ****** peername="YOURIAX2PEERNAME" # ****** date >> slap.log echo "Testing $peername peer" >> slap.log /usr/sbin/asterisk -rx 'iax2 show peers' |grep -i $peername >> slap.log /usr/sbin/asterisk -rx 'iax2 show peers' |grep -i $peername > reg_status sleep 1 #We then Scan the Status and see if we're online or not... TEST="OK" if grep $TEST reg_status > /dev/null then echo "All OK Here" >> slap.log exit #Abort, we are online, all is well... fi #IF we're this far down, we've lost IAX. Log the incident. echo "we have a problem with $peername, Restarting it" >> slap.log #Restart the IAX2 trunk. Delay required for some reason. /usr/sbin/asterisk -rx 'module unload chan_iax2.so' >> slap.log sleep 90; /usr/sbin/asterisk -rx 'module load chan_iax2.so' > /dev/null echo "Restarted it Now lets check status" >> slap.log sleep 5; /usr/sbin/asterisk -rx 'iax2 show peers' |grep -i $peername >> slap.log #We record the status of the IAX2 Trunk /usr/sbin/asterisk -rx 'iax2 show peers' |grep -i $peername > reg_status sleep 1 #We then Scan the Status and see if we're online or not... TEST="OK" if grep $TEST reg_status > /dev/null then echo "All OK Here" >> slap.log exit #Abort, we are online, all is well... fi #IF we're this far down, we've lost IAX. Log the incident. echo "we have a problem with $peername, Restarting it" >> slap.log #Restart the IAX2 trunk. Delay required for some reason. /usr/sbin/asterisk -rx 'module unload chan_iax2.so' >> slap.log sleep 120; /usr/sbin/asterisk -rx 'module load chan_iax2.so' > /dev/null echo "Restarted it Now lets check status" >> slap.log sleep 5; /usr/sbin/asterisk -rx 'iax2 show peers' |grep -i $peername >> slap.log #We record the status of the IAX2 Trunk
======Code ends======
This seems to do the trick and its cronned to run every night or hour in some cases.
UPDATE
on testing and speaking to suppliers. We would advise adding the following settings to your IAX2 peers
qualifysmoothing=yes qualifyfreqnotok=30000 qualifyfreqok=120000 qualify=yes
With this added we have not seen any unexpected unreachables.
The RTX8630 is complete cordless telephony solution offering a great scalability. The system is expandable and can grow with the business; from 1 to 40 bases and up to 200 users. The RXT8630 offers seamless call handover and repeater support. There is a choice of two different RTX DECT handsets, both with a high quality colour screen and wideband audio.
System features
Pricing:
RTX8630 Base RRP: £189.00+VAT
RTX8430 Handset RRP: £99.00+VAT
RTX8630 Handset RRP: £140.00+VAT
Call for availability and project pricing
We get calls regularly on where to set the callerid in Elastix IPPBX systems. There are 3 places it can be entered for external caller ID and some can overide others but not all. so here is a simple explanation
Firstly, You can set it in the Extension, Trunk and the Route, In the route there is a check box as well.
1. If you set it in the Trunk and no where else it will send this out as the CLI.
2. If you set it in the Trunk and in the Extension it will send out the Extension external cli as the CLI.
3. If you set it in the Route, Extension and the Trunk and don’t tick the override it will send the Routes CLI
4. If you set it in the route and the Extension and the Trunk and tick the override it will send the extensions CLI
5. If you set it in the Route and the Trunk and tick the override it will still send the Routes CLI
I hope that makes sense :-)
This Short Video shows you how to setup custom extensions in Elastix and FreePBX
These can be used for calling mobiles or other external numbers that you want users to be able to dial as if they were extensions.
Not only was Digium the first vendor of telephony interface cards built specifically for Asterisk, but it has always been the market leader, with over 50% of the world’s board business.
Digium analogue telephony cards are high-performance, highly reliable and cost-effective interfaces for POTS lines to your Asterisk solution. Multiple applications can be created to satisfy the business needs of any organization when using Digium analogue cards in concert with Asterisk software, the Linux® operating system and standard PC/server platforms.
Digium’s super-reliable digital line cards connect Asterisk-based communication systems to T1, E1, J1 and ISDN-BRI interfaces.
The Digium Hx8 Series are high-performance, cost-effective hybrid analogue and BRI telephony interface cards providing the capability to seamlessly integrate mixed-mode environments in a single device. Use the telephony card selector to identify the card that fits your requirements.
For configuration and pricing please email or call us. We always want to speak to customers buying cards to make sure that it will be compatible with their server hardware.
If you want to set up timed call flow in Elastix but still have the ability to override for holidays and when the office is open late you have a few extra steps to add.
We will assume you have your queues and extensions setup for this video. If you havent set your extensions up see our other video on setting up extensions.
We have used 2 day/night modes, One at before the call enters the time condition, This means that you can override day service for holidays etc and another at the end that means the call can be forced to go to a night queue instead of voicemail.
I hope you found this useful and keep coming back for more.
UPDATE
We have recently added the 5th system to the customers international VoIP network. This system was for their Polish office and is linked to their Tokyo, Sydney, Singapore and London office systems.
For this site a Sangoma FREEPBX 60 system was chosen for ease of remote deployment and reliability. The Tokyo & Sydney offices already has a Xorcom XR2000 systems whilst the London and Singapore offices have a Openvox Asterisk solutions. .
For the New International offices FreePBX systems were chosen as they provide a full turnkey system that can be sent out to the office plugged in. The systems initially obtain their IP address by DHCP and once a port is forwarded through the firewall to this address a fixed IP address is assigned and the customer firewall updated. Access to The GUI is by a SSH tunnel so that other than a random port for SSH and a port for IAX2 no other ports need to be opened on the customer firewall. Endpoint manager makes the deployment of handsets on the remote systems a simple and reliable process.
All systems have been linked by IAX2 trunks and the dial-plan configured so that desk to desk calls can be made between all offices and outgoing calls break out of the closest geographic system, for example a user in Sydney making a call to a UK number will have the call originate from the London system and the same goes for Tokyo, Singapore and Polish users calling UK or international numbers.
The network of systems is key to the support of the customers 24×7 support service. This is controlled by a dial-plan that is complicated by the fact that Japan does not have “Daylight saving” so even though the calls land on the UK system we had to configure the dial-plan to take into account local time in Tokyo and not base routing solely on UK time. This has proved reliable and very successful.
All systems on the network are monitored 24×7 by our Nagios monitoring platform, Not only monitoring Asterisk but also monitoring the status of the international IAX2 trunks.
The Elmeg IP290 are a clone of the Old Snom 190 Sets. and these did support auto configuration. We hoped that it was a simple change to get the files working with the Elmeg.
It turns out that it wasn’t, There are a few gotchas.
Firstly you need to configure your dhcpd.conf
Add the following to the general section
option snom-setting code 66 = string;
option snom-bootfile code 67 = string;
Then the following to the subnet
class “snom-phones” {
match if substring(hardware,1,3) = 00:09:4f;
option snom-setting “https://SERVERIP”;
}
Theb the following are the two files you need to create .
elmegIP290.htm
<html>
<pre>
# example snom general setting file
# After each setting (before the colon) you can set a flag
# General language and time configuration parameter
language: English
web_language: English
timezone: GBR-0
time_server: pool.ntp.org
ntp_server: pool.ntp.org
date_us_format: off
time_24_format: on
user_host1: SERVERIP
user_host2: SERVERIP
tone_scheme: GBR
</pre>
</html>
elmegIP290-00094FMACADDR.htm
<html>
<pre>
# example snom specific setting file
# After each setting (before the colon) you can set a flag
user_name1: 345
user_pass1: PASSWORD
user_name1: 345
user_realname1: 345
user_host1: SERVERIP
user_srtp1: off
user_dp_str1: !([^#]%2b)#!sip:1@d!d
# You may add up to 4 (snom300/ 12 (snom320,snom360,snom370) accounts
# set 1st account to active outgoing identity
active_line: 1
</pre>
</html>