Using a 4 ports FXO gateway to use our Calling Card system without having DID numbers

Case description; using your own telephone numbers for the use of the calling card system

XeloQ Communications delivers DID numbers in 60+ countries but unfortunately some countries lack numbers or the use of it for Calling Card systems is not allowed.

For the company (our resellers) offering the Calling Card system, using your own, local PSTN lines could also be much cheaper than offering a 0800 Toll Free number. This because all costs for calling to those 0800 numbers will be charged to you, the calling card supplier. This can go up to 35 cents per minute!

That is why we offer a solution using your own PSTN lines connected to a so called FXO gateway. Calls are coming in on the PSTN lines and sent out through the Internet connection using multiple SIP accounts and forwarded to the Calling Card prompt.

Below, you will find an example of deploying this with a ClipComm 4 ports FXO gateway.
You can use any other FXO gateway with 1, 2, 4, 8 or more FXO ports.

Do not forget all calls need to go out through your Internet connection so that must be capable enough to carry all calls. Also, preferably select the g729 codec on the SIP / VoIP ports by default.

Configuration of ClipComm CG 410 FXO gateway (4 ports) for XeloQ’s Calling Card system:

-An Ethernet cable from your switch goes on the WAN port
(and will get an IP address through DHCP; find out what that IP address is)

-the 4 RJ11 cables from the PSTN (analog lines) connect to the CH1 – CH4 ports
(these are the lines you call into with a mobile or normal phone)

-Open browser to (example; but using port 1001 is important)

-Login with: admin / 0000 (default name / password)

-Choose System Configuration / VoIP and configure all 4 ports like the example below
(use 4 SIP accounts for each port it’s own account and use your own SIP server if you are a XeloQ reseller).

Fill out like the screen below:

Setup for PSTN to VoIP forwarding:

-Go to Supplementary Function:

Only configure the Channel PSTN1 – PSTN 4 section; leave the Channel – VoIP1 – VoIP4 as it is.

Click to select ‘immediately’ at • FXO off-Hook on Call Forwarding to VoIP


Fill out like the screen below:

-Leave the rest as default. This is enough to get the FXO gateway to work on the Calling Card system.

Now what happens is this:

Customers call in with a normal / mobile phone to the any of the 4 PSTN lines; then the call gets DIRECTLY forwarded to the VoIP number 909 (which is the calling card prompt).

After that enter Card Number + Pin code (you created Calling Card numbers for your users), then dial out as International number; the corresponding calling card will be charged.
The call from the SIP account to 909 is a free call. The calls to the local PSTN lines, are charged normal local rate (if you use normal analog telephone lines with normal charged numbers of course).

How to configure this ‘Two Stage Dialing’ with other FXO gateways?

Other FXO gateways like Grandstream GXW-4104 / 4108 work in a similar way but configuration can differ. As long as you understand the basics, you can configure any FXO gateway using this example.

Good luck.

This configuration example can also be found on the Support pages on our website; www.XeloQ.eu

Kind regards / Met vriendelijke groet,
XeloQ Communications Support Department

Do you have any clue what a SIP trunk is and what you can do with it?

Most of the companies using IP PBX systems, still use the traditional landlines while you can SIP trunk to XeloQ and save on landline costs, calling costs and have the ability to get telephone-numbers from more than 60 countries delivered to you phone system.

Here are a few of the Certified IP PBX manufacturers that XeloQ can SIP trunk from:

Why isn’t everybody aware of this? Is it lack of knowledge? Do people not trust VoIP connections? Are they scared of it?

Let us know please and post your comment below!

Learn more of saving money and gaining functionalities at our website: http://xeloq.com/#/website/categorie/1722/xeloq-voip-sip-trunk

We also wrote another BLOG on this; check it out: http://xeloq.wordpress.com/2009/11/19/7/

What do you as customer choose? Hosted VoIP or an IP PBX onsite? Please tell us!

Hosted VoIP is nothing new; it is here for several years and also XeloQ Communications delivers Hosted VoIP / Hosted PBX for 1,5 year now.

But is not always suitable for each and every company. There are limitations in bandwidth of your Internet connection, the maximum number of extensions in ‘real-life’, fallback and failover and something called ‘emotion’.

Not every company wants his or her Telephone Exchange hosted in ‘some datacenter’ (although it is a complete secured and backed up environment).
It is basically the same idea of hosting your email and administration / documents. That is just not suited for some companies as well.

For those groups of companies, XeloQ delivers the so called Onsite IP PBX. It runs in your own office and you control the dice.

What is your choice? A Hosted PBX or an IP PBX onsite? And why?

Tell us. We would like to hear from you!

More information: http://www.XeloQ.com

Kind regards,
Sales Team XeloQ Communications

How are we doing? Tell us what you think about XeloQ Communications

Every now and then we launch a small survey to measure our Customer’s experience and Customer’s satisfaction.

So, XeloQ customers all around the world; tell us:   How are we doing?

What do you like about XeloQ? What do you dislike? Why did you come to XeloQ? Were you referred to us or found us yourself on the Internet?

Do you understand what we deliver? Is all clear? If you were our webmaster, would you change our website? Why? What would you change? Did you already talk to our support or customer service? And did you like that? Or not?

Just tell us…..but be polite otherwise your Comment will be removed.

As we do every 6 months (what normally was done by an email survey), we listen to you and try to change or / and improve our services and the way we interact with our customers. We will post replies in this post and on our Twitter channel:  http://twitter.com/XeloQ_VoIP

Forgotten what it is we’re doing? Check it all out at http://www.xeloq.eu

You are important to us so let us know.

Post your comment! This can be in Dutch or English. Thank you very much in advance!
Team XeloQ Communications Internet Telephony

(backtrack to earlier comments on old Blog:  http://xeloq.blogspot.com/2009/11/how-are-we-doing-tell-us-what-you-think.html )

SIP SoftPhone on Mobiles (Mobile VoIP) – which program do you use and why?

As being in the VoIP business for 7,5 years, we now see a shift to use Mobile VoIP. Nothing more than using your (smart)phone to make Mobile VoIP calls and bypass International Roaming.

It works great and saves up to 90% on calling costs which is great!

XeloQ partners with NimBuzz, fring and our tech team found the ‘hidden’ settings of Windows Mobile to use their built in stack.

Check it all out on http://xeloq.com/#/website/categorie/1723/mobile-voip

Also some of our customers already use SIPDroid on their Android phones.

Which Mobile SIP SoftPhone are you using?

Why? What is it advantage over the mentioned ones? Is it free or low cost?
Does it have low bandwidth codec’s like g729 on board? Did you already find a SIP SoftPhone for BlackBerry and Palm OS?

Tell us please. We would like to hear from you.

Zojuist onze Blog vernieuwd met eMail updates; easy as 1-2-3

Dit is interessant.

De Blog wordt gevuld met de inhoud / subject van de email die je vanuit een willekeurige emailclient kan sturen (ja…wel een secret email adres…) en tegelijkertijd wordt er een Twitter tweet geplaatst….op http://twitter.com/XeloQ_VoIP met de subject van de Blog (welke dus het Subject was van de email).

Waar gaan we het over hebben? Over het gemak van het per email vullen van de Blog 😉

Makkelijk of niet? Bruikbaar? Nodig? Waarom?

In ieder geval worden de sociale netwerken, blogs, mails, company-websites, connecties met Linked-In allemaal op een eenvoudige manier gekoppeld.

Laat je comment achter….of niet.
Team XeloQ Communications – www.XeloQ.nl

Why still using PSTN lines while you can use SIP Trunks? (and save up to 90% on your calling costs)

Most of the companies deploying VoIP solutions like IP PBX systems still use an ISDN (E1 / T1) connection to the traditional telephone network to terminate (or originate) their calls.

Why? This is not only more expensive looking purely at the calling rates but also the high investments in landlines should not be overlooked. There is a very good alternative which most people are not aware of.

Bringing a SIP Trunk in place (which is part of any IP PBX like Asterisk, AXEOS, 3CX, pbxnsip, ShoreTel and many other free or commercial IP PBX solutions) saves you directly on all of this.

Savings up to 90% on telephone calls are possible and getting rid of your landlines saves another great sum of money. You can always keep a few of your current ISDN / T1 lines for backup / fallback but like 75% of the current landlines can be terminated.

Now you want to know more about this? Contact XeloQ Communications today and we can tell you exactly how to do this!

Take a closer look at http://xeloq.com/#/website/categorie/1722/xeloq-voip-sip-trunk for more details or email us at sales@xeloq.com

What do you think about this? Feel free to add your comment below!