Archive for the ‘Linux’ Category.

CallWithUs launch UK based SIP server

If you’re based in the UK or Europe and looking for a cheap ITSP (VIOP provider) it might be worth looking at CallWithUS as they’ve recently launched a UK based SIP server.

As well as the US based servers sip.callwithus.com, east.callwithus.com and west.callwithus.com you can now use uk.callwithus.com. I now get sub 6ms pings from my Asterisk server in BlueSquare to the CallWithUs server.

I’ve been using CallWithUs for a while now and they provide very competitively priced DIDs and termination rates.

UK based Trixbox and Elastix Virtual Private Servers

Sysadminman is now offering UK based Virtual Private Servers running Trixbox and Elastix.

Please see here for details - http://sysadminman.net/uk-voip-vps.html

CallCentric trunk setup with Asterisk/FreePBX

Here is my CallCentric configuration for FreePBX.

If you’re thinking about signing up with CallCentric please use my referral link here. Thanks.

Trunk Name: CallCentric

PEER Details:

username=1777XXXXXXX
type=peer
secret=PASSWORD
qualify=yes
nat=no
insecure=very
host=callcentric.com
fromuser=1777XXXXXXX
fromdomain=callcentric.com
dtmfmode=rfc2833
disallow=all
context=custom-get-did-from-sip
canreinvite=yes
allow=ulaw

Register String:

1777XXXXXXX:PASSWORD@callcentric.com/1777XXXXXXX

Please note: the above number starting 1777 is your account number and not you DID number

You also need to add 2 lines to one of the configuration files to correctly extract the DID number from incoming calls.

Edit /etc/asterisk/extensions_custom.conf

Add the following lines

[custom-get-did-from-sip]
exten => _.,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)

Then restart Asterisk

You should now be able to create Inbound Routes based on you CallCentric DID numbers

Sipgatge trunk with Asterisk/FreePBX

I’ve used Sipgate for the past few years with my Asterisk box and have been pretty impressed.

For anyone else looking to use Sipgate with Asterisk/FreePBX here is my trunk setup

Trunk Name: Sipgate

PEER Details:

username=1234567
type=peer
secret=XXXXXXXX
qualify=yes
nat=never
insecure=very
host=sipgate.co.uk
fromuser=1234567
fromdomain=mydomain.com
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=yes
authuser=1234567
allow=ulaw

Register String:

1234567:XXXXXXXX@sipgate.co.uk/1234567

If you are using NAT between your Asterisk box and Sipgate you will need “canreinvite=no” and “nat=yes” or you will probably get one way audio only on your calls

Edit: context changed to “from-trunk” to make the post clearer

UK based Asterisk VPS with FreePBX and A2Billing

Sysadminman is now offering UK based Asterisk VPS servers that have Asterisk, FreePBX and A2Billing installed and ready to use. These VPSs are hosted in one of the premier datacentres in the UK (BlueSquare) and have low pings to UK and the rest of Europe.

Asterisk and FreePBX combined make an extremely flexible and easy to manage “virtual telephone system” and A2Billing is a billing application enabling you to charge for calls made using your system if required.

Some possible uses of an Asterisk VPS are -

  • Small businesses wanting to manage a collection of SIP extensions
  • Reselling calling cards or VOIP termination
  • People who do a lot of traveling and want to stay in touch wherever they are
  • Route calls based on time of day, DDI, CLID, etc…
  • Provide contact telephone numbers in cities around the world all routed to your local phones over the internet
  • Voicemail to E-mail

See here for more details and have you have any questions please get in touch

Matt

Cheap international phone calls with Asterisk

If you make a lot of international phones calls, or even if you don’t make a lot of them but need to be flexible in how and when you make them, running an Asterisk server could be the way to go.

Asterisk is a OpenSource telephone system that runs on top of Linux. Asterisk can be quite difficult to manage as it involves editing text based configuration files. If you don’t want to do this you can install a web based management interface called FreePBX. This provides lots of management functionality from a nice web based GUI.

Here are just some ideas for how you could use your own phone system -

Cheap calls abroad from your cell phone

Asterisk/FreePBX provides a feature called DISA. This enables to dial into your Asterisk server (via a local access number available from an ITSP such as CallWithUs) and then dial back out again (again via an ITSP). This would enable to call anywhere in the would at a local rate, right from your cell phone.

Continue reading ‘Cheap international phone calls with Asterisk’ »

Trying to cancel your fusionxhost account!

For a few months now I have been backing up my server using rsync to some space at fxhbackup which is part of fusionxhost.com. The backups worked well and I never really had any problems. There was an issue with the control panel showing my disk usage incorrectly but I opened a ticket and they didn’t seem too bothered - and I wasn’t either as long as my backups carried on working.

The only issue I really had with the service was trying to cancel it recently. I couldn’t find a link in my control panel, or on their website, to cancel the service so I opened a ticket asking them how to cancel my service. For 2 days nothing happened and then, with no communication, my access stopped working.

I have since e-mailed them asking if the would confirm that, in fact, my service has been terminated but no reply.

So, I can do nothing but cancel my PayPal subscription and assume that that’s the end of it.

A real shame of a way to end what was a reasonably pleasant experience.

I don’t think I’d go back to them. The service worked pretty well but I can do without the hassle of trying to chase them up over account queries.

So, fusionxhost, if you’re listening out there please let me know if my account has been canceled!  :-)

Update: fusionxhost did contact me about a week later to apologize and confirm that my account had indeed been canceled. Who knows, maybe this post even helped  :-)

So, would I recommend them? Maybe. The service was pretty good, I never had any problems with it.

Limit SMTP connections for OpenVZ VPS

I’ve started renting out some OpenVZ VPSs for a few people and wanted to make sure that they couldn’t be used to send spam. One of the easiest ways to do this is just to limit the number of outbound smtp connections allowed from the VPS using iptables.

I used the following iptables rules on the OpenVZ host node to accomplish this -

# Limit number of SMTP connections from Mail Server
iptables -A FORWARD -o eth0 -p tcp -s 77.211.239.14 --dport 25 -m limit --limit 3/minute -m state --state NEW -j ACCEPT
# iptables -A FORWARD -o eth0 -p tcp -s 77.211.239.14 --dport 25 -m state --state NEW -j LOG
iptables -A FORWARD -o eth0 -p tcp -s 77.211.239.14 --dport 25 -m state --state NEW -j DROP

The ip address is the ip address of the VPS. The optional log rule in the middle (that’s commented out) is useful when you are setting this up so you can check that packets are actually begin affected by the rules.

Using callwithus with Asterisk, FreePBX and A2Billing

The instructions below assume that you have got Asterisk, FreePBX and A2Billing installed and working together.

Below are the first steps in setting up a callwithis DID number and passing the call through to A2Billing. This number can then be used as an access number for your calling card clients.

Once you’ve signed up for your callwithus account and purchased your DID number the next thing you want to do is modify how your DID number gets presented to your Asterisk box so you can route it to a2billing.

On the callwithus website, click on DID on the left hand menu and then locate the DID number you want to use and click ‘edit’. Now under ‘DESTINATION’ add ‘/yourdidnumber’ the the end of ‘SIP/youraccountnumber’

This will cause callwithus to append your DID to the call details when it is passed through to your Asterisk server. See the image to the right for details.  

 

 

 

Next we want to go into FreePBX and register A2Billing as a ‘Custom Destination’.

If you don’t have the ‘Custom Destination’ menu in FreePBX you will need to go to the modules menu option and install the ‘Custom Applications’ module.
Enter ‘a2billing’ as the Description and ‘custom-a2billing,${EXTEN},1′ as the custom destination

 

 

Now we want to configure our callwithus trunk.

 

 

 

Configure your trunk as in the images here, obviously using your username and password. More information on how to setup your trunk can be found on the callwithus website here.

 

 

 

Now we just need to configure our inbound route. This inbound route decides what happens to a call when it is presented to your Asterisk server. Give it a meaningful description and then in the ‘DID Number’ box enter your callwithus DID in the same format you used in step one above.

Scroll down to the bottom of the configuration page and you should see the ‘a2billing’ ‘Custom Application’ that you setup earlier. Select this and then click on Submit. Don’t forget to click on ‘Apply Configuration’ at the top of the FreePBX page to get Asterisk to read your changes.

 

And that’s it for now. If you call your callwithus DID it should come to your Asterisk box and then be passed to a2billing. Depending on how you’ve got a2billing configured will determine what happens to that call now.

I’ll discuss in a later post some ideas about how to configure a2billing

Rhapsody, Ubuntu and Firefox 3

If you’re having trouble logging in to the Rhapsody website and you are running Firefox 3 on Ubuntu (or possibly other flavours of Unix) then the following might be useful.

If, every time you try to log in to the Rhapsody website, you are asked to install the Firefox plugin, even after apparently installing it, then the plugin is not getting installed correctly.

To install the plugin manually try -

Start a shell prompt

cd .mozilla/plugins/
wget http://forms.real.com/real/player/download.html?f=unix/rhapx/RhapsodyPlayerEngine_Inst_Linux.xpi&install=rwih
unzip RhapsodyPlayerEngine_Inst_Linux.xpi

Then restart Firefox.

Hopefully, now you should be presented with a username/password box when you try to log in rather than the plugin install prompt.