Free DID Numbers, Free Voip Calls, & more Voip Info

[emoci]

Thanks for the reply, here are the settings and my settings are in bold. I'm new to this, so I don't know what most of this means. I was able to place a call to England today without a problem, but Canada seem to be hit and miss. But using their software from my PC the calls go through no problem.

Service profile: IETF / Nokia 3GPP
Public user name: username@internetcalls.com (obviously my username isn't username)
Try putting just Username here (no @internetcalls.com....do this change last, it's the least likely to be the problem in my opinion)
Use compression: No
Registration: When needed / Always on
Use security: No / Yes

Proxy server
Proxy server address: sip.Internetcalls.com
Realm: internetcalls.com (try sip.InternetCalls.com in here)
User name: username
Password: ****
Allow loose routing: Yes / No (Changing this doesn't seem to make a difference)
Transport type: Auto / UDP / TCP
Port: 5060

Registrar server
Registrar server address: sip.Internetcalls.com
Realm: internetcalls.com (try sip.InternetCalls.com in here)
User name: username
Password: ****
Transport type: Auto / UDP / TCP
Port: 5060

Just read up on InternetCalls, a few more suggestions

I've tried a few numbers including cell phones numbers with no luck. The way the phone works, it always has to register before placing a call, so I'm not sure who to disable inbound sip registration (This is exactly the way it should be, registered just before, rather then always on, although I am now thinking Always On may help you avoid the timeout during registration, but it also means you'll waste more data which depending on what it's costing you, may not be a good thing). I also tried to use Fring (third party software which allows me to use Google Talk, Live Messenger, Skype, ICQ and SIP). With Fring the only setup required for SIP is username, password and proxy, but I couldn't make calls with that either. Fring is really good for making voice calls to other IM clients though.

I also tried FreeCalls.com and it's the same as InternetCalls.com, when placing a call, the phone says register to InternetCalls or FreeCalls, but then it just hangs on the dialing screen for a minute or so then just displays a "timeout" message and stops attempting to dial.

I guess my next step will be VoxAlot.

Thanks for the help.

But you did manage to call UK fine right, so it has worked, or better it is possible for it to work, just seems to have trouble doing so sometimes....(sort of trying to rule out the possibility of a problem with the phone itself)

But definitely, try those suggestions with InternetCalls, then give VoXalot a try....

[TRK9]

Thanks for the reply, again.

I should have mentioned this earlier, if I don't put the @internetcalls.com after my username, it fails to register.

Changing the realm seems to have no impact. I can register without a problem using either realm.

Also, data is via WiFi so it's no extra charge than my internet connection.

I found this information: http://www.allaboutsymbian.com/forum...ad.php?t=59389

According to this I should use the server IP in the proxy and registrar server settings. (this makes no difference, just tested it)

-- A weird thing just happened, I tried to call my other phone using internetcalls but cancelled the call after a few seconds. About a minute later that other phone rang with just "416" in the caller id, and there was no response on the line, just dead air, but the call was active. Since it's a cell phone, it would have disconnected if there was no connection.

[emoci]

-- A weird thing just happened, I tried to call my other phone using internetcalls but cancelled the call after a few seconds. About a minute later that other phone rang with just "416" in the caller id, and there was no response on the line, just dead air, but the call was active. Since it's a cell phone, it would have disconnected if there was no connection.

Well whatever your settings when you did this, you may be doing something right.... the 416 CID is all InternetCalls usually sends, either that or Private/Unknown

[lagos]

This is a very useful thread but I am little bit overwhelmed on the amount of configurations presented here. So, I am wondering if someone can recommend a solution/company for my needs.

FYI, I have a Linksys WRT54GP2 Unlocked router/voip box. It has two lines.

What I want to achieve is the following:

-Line 1: 416 number with either Free calls to US/Canada or with best rates or best monthly/yearly fees.

-Line 2: Istanbul-Turkey number since I want people from Turkey to call me at a local price.

-416 fax capable line(virtual number) that is somehow connected to Line 1, and that will somehow distinguish fax signal to be sent to my fax machine after certain number of .rings.

-I have a Smartphone with WM6 OS, I would like to use my VOIP lines if necessary from a Wifi access point.

I don't know if I am asking too much here but any help is appreciated. Thanks in advance.

Tom

[emoci]

-Line 1: 416 number with either Free calls to US/Canada or with best rates or best monthly/yearly fees.

For this request:
-Look for a DID provider (VBuzzer has the cheapest DIDs but their service is a slightly non-existent, it is however possible to make use their DID with a few work-arounds, Les.net and CallCentric have reasonable DID prices too. When looking into this,check whether you pay for incoming calls, get a certain amount of minutes,or get unlimited minutes. Personally I am using VBuzzer for a 416 DID on a month to month basis. If ability to port number later matters you wanna look elsewhere as VBuzzer does not support this)

-For free Calls to US/Cad look into one of the BetaMax companies (usually costs 10 Euro every 4 months, for 300 free minutes a week to a list of countries. The 10 Euro you pay sits in your account and is used when you call places not on the free list, or go over 300 minutes a week. Personally I have been using InternetCalls.com, it's been smooth so far)

-Look for another Pay As You Go plan to have as backup, and for calls to countries you can't call for free (Look at services like VBuzzer,VoiceStick, CallWithUs.com, CallCentric)

-Get yourself a VoXalot account (start with free VoXBasic at first) and integrate all the providers in there (you can have your VoIP enabled cell phone register onto VoXalot using Wifi too, if you have one).

-Get yourself a PBXes account and follow the Guide to set up call through so you can use VoIP through an access number like a Calling Card

- As a freebie, using the post you can get yourself Free Numbers in the US, UK, Italy

(Eg. My costs for this part come to $60 Cad/year to InternetCalls which can then be used towards actual calls, and $2.28 USD/month to VBuzzer for 416 DID, no contracts )

-Line 2: Istanbul-Turkey number since I want people from Turkey to call me at a local price.

This is matter of finding a provider that gives out Turkish DIDs (gonna cost a bit)

Or get yourself a TPad account (which is free) and go through confirmation. They have an Instabul access number. People call that, then enter a 5 digit extension (http://www.tpad.com/Tpad-Country-Loc...In-Numbers.php)

-416 fax capable line(virtual number) that is somehow connected to Line 1, and that will somehow distinguish fax signal to be sent to my fax machine after certain number of .rings.

Fax is a bit tricky. VoIP and fax are not always compatible. Normally if you have a newer fax machine and ansering machine and hook this up to the VoIP line (line 1), the fax machine can recognize the tone just as the Answering Machine takes the call, interrupt the answering machine and take the fax (TAD mode on the Fax).

This means you would use the same number to receive calls and faxes (or have a second 416 number forwarded to the first). Whether it'll work or not is a bit iffy, it'll require some testing.

There may be one way (I am sort of theorizing here as I have not tested). Get a second 416 number, and point it to a VoipUser account (via SIP URI) which offers Fax2EMail service, hence you'd get your faxes by EMail, but if you consider doing this there will be some testing required to see if it is even possible



One thing, what are you gonna use this for, personal use is ok, but anything that borders bussines use may need some consideration due to the reliability of all this. My 2 cents, start getting familiar with all the services slowly without dumping too much money in one, and see how comfortable you are with the way things work.

[emoci]

Voip Networks/Free Hosted PBX Systems

1.VoXaLot: (www.voxalot.com)

VoXalot can be best described as a service aggregator. In a few words they let you put multiple services together, to create a custom VSP. They are not just a PBX, but rather a network on their own...

Free VoXBasic Service includes:

• Incoming calls from SipBroker Access Numbers
• Incoming calls via SIP URI
• Incoming calls from DIDs forwarded to a SIP URI
• Multiple Outgoing Providers can be registered (this supports most providers, except those that require SIP registration for placing calls)
• Dial Plans or Smart Calling which lets you use the cheapest provider for each destination
• Speed Dial List
• Voice Mail (which can also be accessed from regular phones)
• Automatic ENum lookup
• Automatic Geo lookup for premiumnumbers in the UKand Australia (via e164.org)
• Virtual Toll Free (A small script to let people call you from the Web: VoXalot can call their phones, at your expense, or their SIP URI and connect them to you)
• Import/Export feature for Providers/DialPlans/Speed Dials to move them form one acct to the next
• Can be used with any SoftPhone, ATA, and a number of VoIP enabled mobiles

VoXPremium adds the following extra features:

• Call Forwarding based on Caller ID or Provider and Time of Day(send incoming calls to SIP URIs or Phone numbers, so you can always receive your calls)
• Web CallBack (enter two numbers and the providers to use for each number and place a call, hence using VoXalot form PSTN phones as well)
• VoiceMail on the Web
• Incoming Calls from third party providers via SIP Registration (VoXLite: 1, VoXPro: 5, VoXExtreme: 10)

They also provide the SIPBroker service with 200+ donated access numbers around the world to call Voxalot and over 2000+ other SIP based VoIP Networks (millions of VoIP users) for free...

They are associated with (but do not run) the only ENum service open to the end-user, e164.org


2.PBXes(www.pbxes.com)

They provide a hosted Asterisk based PBX service. Their interface remains faithful to the Asterisk interface in general, although it is much more user friendly. Their services include:

PBXes Free: The Free account notable limits are a 10 GB usage, and 2 simultaneous calls, with a 60 min cutoff, and the unability to post in the support forums.
PBXes Premium: Includes a full on PBX including among others Fax as EMail,a full AutoAttendant and Que system, and CallBack access to your Trunks
PBXes Pro: Is geared towards businesses, providing a hosted PBX which can be customized for various clients, including much of the same features as the Premium acct.

For a breakdown and description see: https://www4.pbxes.com/iptel_details_e.html


3.SipSorcery (www.sipsorcery.com)

MySipSwitch is a PBX service concerned solely with Call Routing. In avoiding the hadling of the actual audio stream their application runs much lighter. It is currently an R&D base being run by two technicians and sponsored by BlueFace.ie It provides the following fetaures:

• Multiple outgoing SIP Providers
• Outgoing Dial Plans
• Multiple Incoming SIP Providers
• SIP Registration unto a third party destination URI
• A Call Routing facility handled via DIAL PLANS that has lately added support for multiple simultaneous call forwarding

Their service is new, and the interface is a bit raw, but their outlook looks promising, and used in conjuction with other PBX services, can contribute to a solid VoIP arrangement...

For more info see: http://www.mysipswitch.com/forum/viewtopic.php?t=139


4. LiberaILVoip (www.LiberaIlVoip.it)

Their website is in italian. They provide the following services:

• Incoming and Outgoing Provider Registration
• Dialing Rules
• Incoming Call Filters and Call Forwarding
• VoiceMail
• Gateway Access (CallThrough)
• Call Back access to your provider rates (the only PBX service to offer this for free to my knowledge)

*Some limitations apply, will update when I have a better understanding of what they are


5. GTalk2Voip (http://www.gtalk2voip.com/)
-Used in conjuction with MSN, Yahoo, GoogleTalk messengers
-You create an account by simply entering your email in the home page
-service@gtalk2voip.com (or something similar) adds you as a friend.
-Open a Convo and type: MYPAGE
-Click on the link for your account page
-To CALL type commands on convo with service@gtalk2voip.com :
SIP URI:

CALL user@provider.com

MSN:
If email is username@hotmail.com
then type

CALL username_at_hotmail.com@msn.gtalk2voip.com

If email is username@msn.com
then type

CALL username_at_msn.com@msn.gtalk2voip.com

GoogleTalk:
If email is username@gmail.com
then type:

CALL username_at_gmail.com@gtalk.gtalk2voip.com

If email is username@DomainWithGoogleApps.com
then type:

CALL username_at_DomainWithGoogleApps.com@gtalk.gtalk2voip.com

Yahoo
If email username@yahoo.com
then type:

CALL username_at_yahoo.com@yahoo.gtalk2voip.com

PSTN
Type:

CALL 14161112323

(always type number in international format)
PSTN calls are not free, but you can now add your own providers to GTalk2Voip, and the call will be routed through them. Otherwise GTalk2Voip routes through its own providers and requires you have a balance
-When calling someone for the first time using the commands above, chances are it will not go through, but they will receive a notice that service@gtalk2voip.com has added them as a friend, if they accept, the second time you try they will receive a call from service@gtalk2voip.com
-This works much smoother in GoogleTalk, rather than MSN (have not tested with Yahoo)
-They assign you a number *018xxxxx, that can be reached by dialing SipBroker Access Number + *018xxxxx
or *018xxxxx@sipbroker.com (SIP URI calling), in all this cases you'll receive a call on your MSN/Yahoo/GTalk messenger
-You can also get a DID number, forward it to *018xxxxx@sipbroker.com, and your MSN/GTALK/YAHOO messenger will ring when someone calls you
-There is a generic Voicemail if you don't pick up, or you can forward to a SIP URI and receive calls to your current provider
-You can use them to IM people not using same messenger (http://www.gtalk2voip.com/gtalk_service_im.shtml)
-You can create your own WebCall button (it calls you via Voip, plus uses one of your providers to call the other party and connect you together, if someone enter an actual number though, you will end up paying for the call, but at VoIP rates) (http://www.gtalk2voip.com/webcall.shtml)
-In Summary, neat to play around with. It is a raw framework for now, not very user friendly, but looks promising.
-Using GTalk2Voip with your own Domain: http://www.gtalk2voip.com/forum/topic_show.pl?tid=41



Continued

[cipra]

http://www.voipwise.com/en/index.html

Unlimited free calls to

Australia
Austria
Belgium
Canada
Czech Republic
Denmark
Estonia
Finland
France
Germany
Greece
Greenland
Hungary
Iceland
Ireland
Italy
Japan
Latvia
Liechtenstein
Luxembourg
Malta
Netherlands
New Zealand
Norway
Portugal
Singapore
Slovak Republic
Slovenia
South Korea
Spain
Sweden
Switzerland
Turkey
United Kingdom
United States (+mobile)

Use deepfreeze and use the software even after using up 3 free accounts.

[emoci]

Could you expand, I haven't heard anything of 3 free accounts.

-If you download and just use (no payment) you get calls that are (maybe not) cut off at 1 minute.
New users can try Voipwise out for free for a total of 60 minutes. During this trial period you can only call the destinations marked as free. Register your account by buying credit in order to extend your free calls.

-If you top up once at 10 Euro, for the next 120 days, you get 300 min of free calls to that list of countries per week

What's up with the 3 free account limit and deepfreeze?

http://www.voipwise.com/en/index.html

Unlimited free calls to

Australia
Austria
Belgium
Canada
Czech Republic
Denmark
Estonia
Finland
France
Germany
Greece
Greenland
Hungary
Iceland
Ireland
Italy
Japan
Latvia
Liechtenstein
Luxembourg
Malta
Netherlands
New Zealand
Norway
Portugal
Singapore
Slovak Republic
Slovenia
South Korea
Spain
Sweden
Switzerland
Turkey
United Kingdom
United States (+mobile)

Use deepfreeze and use the software even after using up 3 free accounts.

[cipra]

You can create 3 different accounts [no email activation ] each account allows for 60 mins each. When you go to create the 4th account there will be an error. According to vinay and others the only solution is to use deepfreeze and revert back to state before creation of the 1st account.

[morglum82]

Well you should have these details from ACANAC (more likely if you have your own ATA):

1. User
2. Pass
3. Proxy/Server ( which I assume is one of these eastern.acanac.com eastern.acanac.com, eastern.acanac.net, eastern.acanac.net, eastern2.acanac.net )

Now you can use that info to try a registration on PBXes or MySipSwitch. If it works this is the idea:

-Calls to ACANAC Number go to MySipSwitch and/or PBXes
-Calls to SipBroker PSTN then SipCode (MySipSwitch and/or PBXes sipcode) followed by number would let people reach you on the same phone fromm around the world.
-Your Device which is hopefully unlocked will be registered with either PBXes or MySipSwitch (or for better results VoXalot for $15/year)

Sorry, can't give you better news...it's quite a bit of work

Thanks emoci for all this help, butbeign a voip newbie there's so many acronyms/names I cant seem to figure out when trying to setup PBXES.

Here's what I have:
a voxalot account (this guy can will be reached through SIP broker access numbers)
a PBXES.com account (my understanding is that this guy will receive calls to voxalot and then forward them to my ACANAC number, but how?)
and an ACANAC account to which my SPA3102 is registered.

Can you be more specific about how I should set up PBXES.com? I'm not sure what to do with extensions / trunks / inbound outbound routing (well. everything!!)


Thanks!

[morglum82]

nevermind, i stated using mysipswitch and its MUCH easier than pbxes.com..


now if only SIP BROKER's circuit werent always full I could test it..

[emoci]

Thanks emoci for all this help, butbeign a voip newbie there's so many acronyms/names I cant seem to figure out when trying to setup PBXES.

Here's what I have:
a voxalot account (this guy can will be reached through SIP broker access numbers)
a PBXES.com account (my understanding is that this guy will receive calls to voxalot and then forward them to my ACANAC number, but how?)
and an ACANAC account to which my SPA3102 is registered.

Can you be more specific about how I should set up PBXES.com? I'm not sure what to do with extensions / trunks / inbound outbound routing (well. everything!!)


Thanks!

If you know those settings on the SPA 3102 (it's an unlocked SPA 3102 right?) you can make this happen!

See http://www.scopezoom.com/guide6.htm or PM me if you need some help!

[emoci]

nevermind, i stated using mysipswitch and its MUCH easier than pbxes.com..


now if only SIP BROKER's circuit werent always full I could test it..

Edit: OK, I am confusing things.....you are trying to set up so you can receive calls,but you had an issue with PhoneGnome correct?

Try this (Your SPA has more than one line right?):

To receive Calls:
-In your SPA set the Line registered with ACANAC to forward all calls to xxxxxx@voxalot.com
-Register another line of your SPA with that VoXalot Account, and hook a phone here
-When someone calls ACANAC number, or SipBroker + *010xxxxxx (xxxxxx being VoXalot number), if everything is working your phone should ring

ACANAC (SPA Line 1)-----forwards to------VoXalot (SPA Line 2)-------Rings Phone

To Make Calls (from your VoIP line)
-Either configure ACANAC as an outgoing provider in VoXalot, or since you have a SPA 3102, you can probably set it up as a gateway on SPA

VoXalot (Line 2)----places call------Gateway or VoXalot Outbound Provider--------Rings Phone

To use ACANAC to make calls from your Cell/Other Phones as if it was a Calling Card:
http://www.scopezoom.com/guide6.htm

Keep me updated, I'm kinda curious if you manage to get this working with ACANAC

[morglum82]

edit: never mind, i got it 90% working

I cant use mysipswitch to forward between VOXALOT and ACANAC.
I can use mysipswitch to forward voxalot#1 to voxalot#2.

What I ended up doing:

-mysipswitch takes calls from acanac and voxalot and forwards them to myusername@213.200.94.182 (my mysipswitch user)
-mysipswitch dialplan is as follow :

exten => _X.,1,Switch(user,pass,${EXTEN}@acanacsip)
that way I dont need to dial a prefix when I pick up the phone.


Now for the missing 10%:
I cant access my voicemail anymore. Normally I'd do *123 (its what acanac wants it to be, and that's what I've set my 'blind transfer code' on the spa3102 to. But now when I dial *123 I get a busy signal. Any idea how to fix that?

Thanks!!!
Simon

[emoci]

Hey emoci,

I've been going nowhere all day again

BAsically, the SPA3102 only allows to receive inbound SIP from 1 provider.

(1) So I have to either log in to ACANAC and have voxalot forwarded to it using mysipswitch.

(2) Or I login into mysipswitch and have mysipswitch register to botch acanac and voxalot

(3) or i login into voxalot and have mysipswitch forward acanac to voxalot.

Well, none of the the 3 options worked.

I cant seem to have anything forwared to/from acanac, even though mysipswitch does register to it.

I confirm that mysipswith can forward stuff : I've had calls routed from VOXALOT1 to VOXALOT2 and that worked.

...
guess I'll just never mind
thanks though!

What ATA allow to register 2+ inbound SIP ?

Well, you definitely have tried all the possible options. Under monitoring at MySipSwitch you may see some messages to at least give you an idea what's happening as you try these...

PAP2 has 2 ports