VoipDiscount.com: Free Asterisk Calling Returns to the VoIP Wild West

We return to the VoIP Wild West again today with another free calling discovery. As most of you know, these don’t last forever, but they sure are great while they exist. VoipDiscount.com is one of the better deals to come along in quite a while because of the scope of the countries to which you can place free calls. As you can see from their banner, the list includes not only the United States and Canada, but also 50 49 OTHER COUNTRIES including Mexico and most of Europe, South America and the Far East! And calls to formerly expensive places such as Costa Rica are about a penny a minute. Click on the banner for their complete price schedule. The only (published) catch is you have to open an account with at least $15 to cover the upfront costs of calls to these other places. Be sure to read the comments for some others!

Getting Started. To begin, you’ll need to download their softphone client from the VoipDiscount.com web site using a Windows PC. When you’ve installed the softphone, run the application and create a username and password for yourself. Once you’ve verified your email address, you can add 10 euros (about $15) to your account using a major credit card or PayPal, and you’re all set. Be sure you configure the client to NOT start up each time you start Windows since we’re going to be using the service with your Asterisk® server. And, when you shut down the client, be sure to look in the Windows Toolbar at the bottom of your display and REALLY exit from the softphone to shut it down. Otherwise, you won’t be able to connect to the service with Asterisk since you can’t have two clients logged in at the same time.

Creating a VoipDiscount Trunk. If you’re using Asterisk@Home or the Asterisk Management Portal (AMP), or FreePBX, then getting things set up for VoipDiscount is a breeze. Open the management portal with your web browser pointed to the IP address of your Asterisk system and choose Setup->Trunks. Now create a SIP trunk for the service with a Trunk Name of voipdiscount and with Outgoing Dialing Rules in the U.S. that look like this:

001+NXXNXXXXXX
00+1NXXNXXXXXX
001NXXNXXXXXX

Add the following PEER Details in Outgoing Settings using your own username (in three places!) and password. Leave the Incoming Settings blank.

allow=ulaw&alaw
authuser=yourusername
disallow=all
fromdomain=sipdiscount.com
fromuser=yourusername
host=sip.sipdiscount.com
insecure=very
nat=yes
qualify=yes
secret=yourpassword
sendrpid=yes
type=peer
username=yourusername

For the Registration String, enter the following using your own username and password:

yourusername:yourpassword@sip.sipdiscount.com

Click the Submit Changes button and click the Red Bar to update Asterisk.

Creating a VoipDiscount Outbound Route. Now choose the Outbound Routes option and click Add New Route. Fill in the blanks with something like the following:

  • Route Name: OutVoipDisc
  • Route Password: leave blank
  • Dial Patterns: 88|NXXNXXXXXX
  • Trunk Sequence: sip/voipdiscount
  • Now click the Add button and then the Submit Changes button. Then move the Dialing Route for OutVoipDisc to the top of your list of dialing priorities, and click the Red Bar to reload your new Asterisk settings.

    Making a Test Call. Once you complete the above steps, you should be able to make a test call to anywhere in the U.S. by dialing 88-678-1234567 where 88 is the dialing prefix you set up in the Outbound Route (make it anything you like!), 678 is the area code for the call, and 1234567 is the local phone number. For those in other countries or, if those in the U.S. wish to call outside the U.S., add the appropriate dial patterns for each country to both the Trunk Dialing Rules and the Outbound Route Dial Patterns. For example, for U.K. calls, the trunk rule would be 0044. and the outbound dial pattern would be 88|0044.! Dont’ forget the trailing period at the end of both of these entries. Enjoy!

    The Bait and Switch Begins. Well, it didn’t take long. It’s been a day, and Mexico already has disappeared from the free list. It’s now 6¢ a minute. That’s 40%-500% more than Voxee.com depending upon the city you call in Mexico. We’ll keep you posted with new “developments” each week. In the meantime, talk fast and check for current rates before you dial. This is going to be fun!


    Adding Automatic Callbacks to Your Asterisk Voice Mail System. Don’t you just love those Baby Bell phone messages that say “Press 1 and, for a charge of 75¢, we will place this call for you”? Well, now you can add similar functionality to your Asterisk Voicemail System minus the 75¢ charge. Sometimes it’s a lot more convenient to have the computer do the dialing after you’ve listened to a voicemail message particularly when you’re zipping down the highway at warp speed. To add the functionality to Asterisk using the Asterisk Management Panel (AMP) or freePBX, here’s how. Open the settings for the voicemail extension you wish to configure with this option: Settings->Extensions->ext#. Now scroll down to the vm options field and add the following: callback=from-internal. Save your changes and click the big Red Bar to update Asterisk. In the future, when you listen to a voicemail message on this extension and want to automatically return the call, choose 3 for Advanced Options and then 2 to Return the Call. Just be sure your Default Asterisk Outbound Route is configured to dial using the same number format as your received CallerID numbers, and you’re all set. And, by the way, this works with Asterisk@Home versions at least as far back as 1.5. Too bad no one ever bothered to document it. Very slick!


    Nerd Vittles User Map. Thanks for visiting! We hope you’ll take a second and add yourself to our Frappr World Map compliments of Google. In making your entry, you can choose an icon: guy, gal, nerd, or geek. For those that don’t know the difference in the last two, here’s the best definition we’ve found: “a nerd is very similar to a geek, but with more RAM and a faster modem.” The map still isn’t quite representative of where all of our visitors are coming from, but we’re getting there. Give it another week or two. Finally, we’re always looking for the best BBQ joints on the planet. So, if you know of one, add it to the map while you’re visiting as well. We’ll check it out one of these days.

    Want More Projects? For a complete catalog of all our previous Asterisk projects, click here. For the most recent articles, click here and just scroll down the page.

    Coming Attractions. By the end of this summer, we’ll be ready to begin tackling some of the Asterisk hardware cards including the new Sangoma cards and the soon-to-be-released Digium cards. All of these new cards now support hardware echo cancellation. There’s just one catch: we don’t have any. If you’d like to make a small contribution to assist with the project, just click on the donate button in the top right corner of the main page. When the kitty grows sufficiently to fund the cards, we’ll begin our in-depth tutorials to assist everyone in making the right hardware choices. And, by the way, we’re not talking about huge individual donations. We’d much prefer 200 people donating $5 each rather than Bill Gates coughing up $1,000. But, heck, whatever works to get the hardware funded.

    Headline News for the Busy Executive and the Lazy Loafer. Get your Headline News the easy way: Planet Asterisk, Planet Gadget, Planet Mac, and Planet Daily. Quick read, no fluff.

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    38 Responses to “VoipDiscount.com: Free Asterisk Calling Returns to the VoIP Wild West”

    1. Williamts99 says:

      Just something to watch out for, they are the same company behind VoipBuster, VoipStunt, and also SparVoip. VoipBuster started with 1 euro for the unlimited free calling, then VoipStunt was 5 euro to start, not sure about SparVoip, they cater to speakers of the German language, but now VoipDiscount which from the article is about 10 euro to start. They all follow the same practice so enjoy it while you can. They will start cutting back the countries shortly after they get their burst of users to sign up. Another thing to watch out for is the fact that your credit will expire even if you are using it for free and for paid calling. And the last thing from their FAQ that I don’t have experence with, but it states that if you go over what they define as “fair use” that they will start to charge you for your “free” calls. Not exactly sure what they consider “fair use” but it is something to watch out for. As far as voipbuster, I was using them for a while and it was great until a lot of people signed up, then it would take 10 or so tries to connect a call. Then they started to drop countries from the list, the most popular countries. And then as I was using up my credit I recieved an email stating that my credit expired. Not exactly sure on the time frame, but your credit will expire even if you don’t use it only for free calling. Just a heads up on the situation.

      Will
      Here are the links so you can compare(I think there was one more but I can’t remember it):
      http://www.voipbuster.com/en/index.html
      http://www.voipstunt.com/en/index.html
      http://www.sparvoip.de/de/index.html
      http://www.voipdiscount.com/en/index.html

      [WM: You are correct. We've received a number of comments that are similar, and I tried to caution that these freebies "don't last forever." Just go in with your eyes wide open and enjoy it while you can.]

    2. Maxim says:

      at this time actually you can use their service without even paying the ?¢‚Äö¬¨15 they want you to. So enjoy while it lasts.

      [WM: True, but calls are limited to one minute. Talk fast!]

    3. Jonathan says:

      Re: Adding Automatic Callbacks to Your Asterisk Voice Mail System.

      Works a treat on a@h 1.3 too.

    4. Mike says:

      Ward, Do you know if Matthew, at goiax, will ever start allowing calls (other than toll free) again?? That was a sweeet service at a sweeeet price.

      [WM: I don't think so. I had a conversation with him about how to protect against war dialers, but he didn't seem very interested. Too bad.]

    5. Paulo Durano says:

      I followed step by step voipdiscount instructions,Thank you. AAH shows “registered”. When I dialed 88XXX1234567 it says All circuit are busy now? Thus it mean this is a voipdiscount problem or my AAH installation?

      [WM: Load the Asterisk CLI and see what it says. Is the dial string being sent in the format 001NXXNXXXXXX? Be sure there's not a typo in either your username or password entries. Good luck!]

    6. Been using sipdiscount for over a month now (same as voipdiscount) with no problems on our production Asterisk system as one of the main outbound trunks. There are sites available with dial plans for the countries which are free so that Asterisk uses it for the free locations.

    7. Randy Powell says:

      Ward, love the vittles. I followed directions paid my 10euros and got voipdiscount working great (with one exception). How do I get it to recognize DTMF’s after connect (for IVR systems)? Works with telasip, but not voipdiscount.

      [WM: Add a dtmfmode line to your setup. There are only four choices: dtmfmode=rfc2833, dtmfmode=inband, dtmfmode=info, and dtmfmode=auto. Let us know which one works for you.]

    8. hfdill says:

      voipbuster.com, voipstunt.com, sparvoip.de, voipdiscount.com have all been set up by Finerea S.A. of Geneva, Switzerland, but seem to have been taken over by Betamax GmbH of Germany a couple of months ago (that’s also where most of their DNS servers point to). The Euro 10 fee is good for a limited time after which you need to increase the deposit, unless of course you are using the money to make calls to ‘paid’ destinations. the validity for the deposit seems to be 3-4 months. i have been using them for some time and never had a problem. they are the best deal out there, even if you have to sign up to 2 or 3 of them to cover your ‘free’ destinations.

    9. nick moore says:

      Added to your list….
      http://www.voipcheap.com/
      http://www.voipbuster.com/en/index.html
      http://www.voipstunt.com/en/index.html
      http://www.sparvoip.de/de/index.html
      http://www.voipdiscount.com/en/index.html

      These are all the same company – finarea of switzerland

      I’ve used Voipcheap for over a year now and wondered how the business plan works to be sustainable. Basically it’s ?Ǭ£10 per 6 month’s usage and there are fair usage penalties where the free to POTS calls start to get charged at half a cent a minute. This still gets to be very good value for money especially if you get your mates to use your account from their obscure location back to the states or the UK for free – Remember that the internet initiated call doesn’t care where you are calling FROM ;)

      I’d like to know whether I can use sipura or other ATA hardware phone adapters with this system. fail to see how Vonage et al will remain in business for long.

    10. Nick Moore says:

      It seems that from the noise on some messageboards Tiscali are giving voipcheap and similar users a rough time as they have their own proprietary system. Unfortunately for tiscali users have been voting with their feet.

    11. Randy Powell says:

      Ward, was odd issue with the DTMF’s, worked Telasip fine how I had it, but for the Voipdiscount, I had to change my Grandstream to use “In-audio” instead of “RTP”. Works great now … THANKS again. Also – I know you must have played with A@H2.8 beta 1 with the new freePBX – I LOVE IT (just remember that AA-1 will now be IVR-1).

    12. About the DTMF, at least for the VoipStunt and VoipBuster account that I use, it needs to be set to dtmfmode=inband within the trunk settings. I hope this “Almost free ride” keeps going for some time! LOL

    13. marc says:

      VoipDiscount does not seem to pass along outgoing caller ID information.
      When I make a call, the receiver does not see the Caller ID information
      I specified in AAH. Any suggestions on how to correct this?

    14. http://www.internetcalls.com is another URL see the VOIP wiki for a full list an buisness explanations on how they do this …..

      I got them all working as unmonitored peers, would be nice if someone could let me know how you can get their DDI on these numbers to work, the register string dont work for me as sip show registry does not show any entries after adding them a peer and then a context to dial out works fine also all have a look ar http://www.voipuser.org for some more free trunks …. gl

    15. The callerid is the ddi you can obtain this is what my users see as callerid when using the voipstunt ones … we need to do the incoming bit for that now an example should be provided on sipdiscount.com as this is the only sip only no client software for those accounts offered form Finanarea. Sorry for some of the typos.

      Charlie

    16. Regsitering and using the Firefly softphone or even X-lite as a client is possible as well IAX peering for voipbuster works for me too.

      Been using these all for over a year, I paid the samll fees all to get the DDI and caps removed. Not expenisve!!

      Good luck and have fun ….
      Charlie

    17. Paul says:

      Ward – Superlative tutorials and great ideas! Thanks

      Charles – To get your voip-in number working check out this thread . These settings can be adapted for all the finarea family – sipdiscount, voipstunt, internetcalls, voipbuster, voipdiscount etc

    18. Rob says:

      Can someone help with context for auto callback? I can’t get it to work as my trunk needs a “1″ in front of the number but my outbound dial plan doesn’t. Is there a solution?

    19. Mohidin says:

      hi guys! great job you are doing! Does anyone know how i can use my Cisco ATA186 with any of the Finerea links? They give the settings there but tried with no success. Please help. Thnx

    20. NerdBob says:

      The funny thing is, I posted a long comment a while back, which I believe WM was right in not allowing to appear because of its length but had the same issue. Had it been trimmed and skimmed and posted, it would have given the same message :) There is something wrong with VoIPDiscount & SIPDiscount. Wasted about 24 bucks with these, and now trying to get it back somehow, have written to the address on the invoice. Cant call using their client, Cant call using a Hard Phone, Cant call using an asterisk trunk, no matter what I tried. A confirmation further would be nice.

      [WM: All I can tell you is "mine works" with the settings shown in the article. We really can't debug problems in the blog format. If you're still having problems, post something on Voxilla, and we (and others) will have a look.]

    21. Alex says:

      Enjoyed the free ride for 12hrs:31min:30sec since 4/26/06. Now I can’t make any free phone calls although their website now states: “* Max 20hrs per month of free calls. Unused free minutes cannot be taken to the following month(s).” I’ve written to them to no avail. Calls that have a fee go through with no problem.

      I started using A@H2.8 with your A@H2.7 instructions (used to have an asterisk source based installation) and got it working just fine. Following your A@H2.8 instructions on a working 2.8 requires some tweaking (creating/recreating db tables) and found that some of the Tools (phpmyadmin, asterisk info) went missing in the FreePBX interface.

      Thanks for the GREAT documentation and tips.

    22. John says:

      OK, I have been using their services for months and suddenly, the payment link disappeared and I cant make new payments…sent several emails and post on forums but no replies. Asked couple of friends and realised that some folks can still make payments and others cant…does any one know what is wrong?

      Tried registering a new ID on a new service http://www.voipcheap.com but unfortunately, I cant…does anyone know how to solve this problem, I really want to buy credits and use their service for calling paid destinations?

      thnx

    23. Alex says:

      Updates on the voipdiscount freebie.. New usage policy:

      “Max 300 minutes per week of free calls, measured over the last 7 days. Unused free minutes cannot be taken to the following week(s).
      If limit is exceeded, a minimal rate of 1ct/min will be charged (as shown in the VoipDiscount software).”

      The price of the calls that are charged after the free time is up is very cheap, for Colombia (South America) it’s 1 eurocent a min (US$0.012c/min) which still beats anything out there for Colombia.

    24. Randy Powell says:

      VoipDiscount / VoipCheap no longer have the link for payment with Credit card for me :( Also, I had a client calling India Mobile and they did 4 calls that I didn’t connect that would run concurrent that got billed 1 hour each for. I hope they work it out – was pretty good quality cheap carrier for me till I can do enough volume to qualify for larger carrier status.

    25. Hans says:

      Thanks for this explanation! Actually you are telling there are four programs. But there are 10 all of the same company. Betamax in Germany.
      Here you see the programs and the tarifs with links to their websites; http://www.voipproducts.eu

    26. Alex says:

      They dropped Brazil from their list of free phone calls. Also, I’ve noticed that the 300 minutes seems to be per destination.

    27. ward says:

      Gotcha reminder: ‘Free’ calls now are limited to 300 minutes during any consecutive 7-day period. That’s less than 45 free minutes a day for you math wizards. After that, calls to the free destinations are a penny a minute. And remember, you have to invest another $15 every three months to keep your ‘free’ account active. That makes the so-called free minutes just under a half cent each assuming you don’t use your $15 contribution to call other locations during each three month period. And, no, free minutes do not roll over to the next 7 day period.

    28. Alex says:

      Running Trixbox 1.2.3 FreePBX 2.1.3 and have problems with callback.
      Selecting “2″ in VM, system verifyes the call back number, but never dials out and hangs up my line in a few second.
      Any one has or had the same problem?

      [WM: Read the CLI output and see where the call is failing. Then post the problem on the TrixBox forums if you need assistance.]

    29. Rob says:

      I have been using voxbone IN bound with DID all you can eat for 8 per month 2 channels (business) for a Canadian DID work great no jitter or lag so far I have been use voip discount for a while now too stumbled on it quite a while ago and signed up for it some time after that seems to work for the amount of calling I do however there is some lag.

    30. i think you make people very happy some time one call can save one some body life.thank you for making people talking free its very important what you dowing.thanks frome Dr.obaji

    31. Bruce says:

      Hey, a possible project for your prodigious programmers- I’ve been trying to do an automatic emergency call for the health department where I work. A call to an extension number would initiate an automatic 911 call and play a message when answered. I got asterisk to do the dialout and play a message just with the dialplan, but the message was very garbled,I think because the inintiating phone was bridged to the 911 call. I plan to put the emergency button on a phone’s programmable buttons,to call that extension. A program to move a phone message to asterisk’s outbox folder would be much slicker.

    32. Nailia says:

      Hello! Can someone help me. I have been using VoipDiscount service for one year, but now I have problems making calls. When I make a call, the receiver does not hear me. Does anyone know how to solve this problem. I really want to use their servise. Thank you. Best wishes, Nailia.

    33. piglet says:

      It seems that it isn’t possible to prevent VoipDiscount from starting automatically. There is no option in the Settings to opt against automatic startup, and even when you disable it using Windows Defender, next time you use the program it will be registered for startup again. I regard this is inacceptable practice and I will stop using it as soon as my credit is used up.

    34. chris says:

      Hello to all of you .. my name is chris and i am from greece,I just instaled a wireless router dray tek vigor 2700 series with voip over ip so i can youse my home telephone to make phone calls with the voip discount eaven if my pc is closed.. but my problem is that when i make the calls pepole hear me very low and braking up not clear voice and not loud but i can hear them just fine if any body can help please e mail me at chrismagic@mail.gr thank you a lot, and happy new year to all of you..

    35. Peter says:

      RE: VoipDiscount does not seem to pass along outgoing caller ID information.

      For VOIPstunt etc… – you will need to download their client application and follow the instructions in the settings tab to verify your phone number.
      Once you have done this, leave the outgoing caller id in freepbx blank but add/change the fromuser= to be your validated caller id in 0044 form. Leaving authuser= to be your account login. I have this working from the UK with my sipgate DID and BT POTS number.

    36. vipin sahu says:

      For pc to mobile configuration using asterisk refer this tutorial
      http://vipinkrsahu.blogspot.com/search/label/Asterisk

    37. Abhishek Kumar Srivastava says:

      Nice One! Exactly what I was looking for..

    38. James says:

      No one show try this it is trap for getting credentials from voipdiscount

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