A good percentage of people use Skype for making International calls. I used to do that long time back too (in 2006-7). But then I came across a cheaper and better alternative i.e. SIP or VOIP calling! This is in my opinion the cheapest and better way to make calls. I have used ActionVoip, Freecall and Xeloq but the service I am currently using is Siptraffic. I will recommend Siptraffic to everyone as they have very competitive rates. The only downside is that you need to recharge with a minimum of EUR 200
DISCLAIMER : Before proceeding further, I would like to point out that this method works for me when I use Siptraffic. So please try other SIP providers and check out if they work.
Now back to the topic
If you frequently make SIP calls, you must have noticed that when you call someone, it comes as a “no number” on their display screen or some garbage number like +4156 or +5166 etc.
The method I am going to describe below will let you forward your caller id to the receiver!
Method :
In order to start you need to first create an account at Pbxes. Its a free service (Please note if your usage is above 2000 mins/month, they will block your account and you need to purchase a premium package for a month at least!)
Once your account is created, login to your account.
Then click on Extensions:

Then click on “Add extension” and select “SIP” :

In the above, you just need to fill the following :
Extension Number: << A unique number to login to this extension. If you user id is abhitest then to login to this extension you need to use abhitest-100 (if you used 100 as extension number) >>
Display Name: << A display name to indentify this extension >>
Password : << Put password of your choice >>
Important : Click Submit and click the red bar which comes on the top to apply changes!
Next click on “Trunks -> Add SIP Trunk ” :

Here comes the “tricky” part where you need to give the caller id. Please fill in the details only in the columns circled above. I will detail what values need to be provided as below :
Trunk name : << A readable name of the trunk for you to identify >>
Username : << LEAVE IT BLANK!!!>>
Password : << Be careful!! In this filed you need to fill in the Siptraffic username and password in the format as “password:username” (without quotes) >>
SIP server or proxy : << Fill in the Sip server IP or address; here its sip.siptraffic.com >>
register : << Since I use only outbound calls, I selected “no(just outbound calls)” >>
Outbound caller ID : << IMPORTANT! This is where you fill in your caller id you want to forward >>
Outbound dialer prefix : << If you use a dialer prefix, you can use here. Otherwise leave it blank>>
Important : Click Submit and click the red bar which comes on the top to apply changes!
Next comes the routing or in simple words, you want your calls to be routed somewhere, so do it here ![]()
Click on “Outbound Routing -> Add route “. Give a name for “Route name” and select the trunk you just created and click add. Leave other options as-is and then “Submit changes”.

Important : Click Submit and click the red bar which comes on the top to apply changes!
Now you are all done!!!!!!
Give a few mins or an hour for changes to take affect. Now in your SIP device (I use my nokia or X-lite on windows) configure the username as your Pbxes username-extension ( e.g abhishek-100). The password will be your extension password as you specified above. Your SIP registrar server will be pbxes.org
Once you have made these changes, start making the calls. The receiver will definitely get the caller ID which you have specified.
Please note, there are too many settings in Pbxes to play with, but I will suggest to start with the minimum and then go beyond imagination
If you liked my post, please provide some feedback, requests or suggestions about my blog here!
-Abhi




Hope you liked the post from 






Abhishek Reply:
March 9th, 2011 at 2:16 pm
Hi there!
I am sorry I was quite busy with my work related things hence could not spend time on it. But I just purchased a EUR 10 action voip account to start testing on it. I have seen some forums which suggest that you can pass the caller ID when you configure a classic extension instead of a SIP one. I am trying and will revert back to you!
-Abhi
[Translate]
Like or Dislike:
0
0