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-   -   HowTo: Accessing VoXaLot From A Regular Phone (https://forum.sipbroker.com/showthread.php?t=985)

martin 02-05-2007 06:31 AM

HowTo: Accessing VoXaLot From A Regular Phone
 
Here are some instructions on howto call your VoXaLot number and voicemail from a regular fixed line or mobile phone.

To call your VoXaLot number:

1. Call one of the SIP Broker PSTN access numbers: www.sipbroker.com/sipbroker/action/pstnNumbers
2. At the voice prompt, dial *010 xxxxxx (*010 is the VoXaLot SIP-Code. See http://www.sipbroker.com/sipbroker/a...iderWhitePages, and xxxxxx is your VoXaLot number)
3. This will ring your VoXaLot phone / device.

To access your voicemail:

1. Call one of the SIP Broker PSTN access numbers (as above).
2. At the voice prompt, dial *010 500 (500 is the voicemail service number. See FAQ)
3. When prompted enter your 6 digit VoXaLot number and pass code.

Note: There are currently over 200 access numbers in over 19 countries around the world.

Hally 02-09-2007 10:21 PM

Sipbroker PSTN numbers for Regional areas
 
Hi Martin, are there any plans to add Sipbroker numbers to regional areas in Australia?

martin 02-10-2007 03:42 AM

Hi Hally,

The numbers are usually sponsored by VSP's. So if there is an Australian VSP that approaches us to sponsor some regional numbers, we will be more than happy to accommodate.

twen 02-11-2007 01:36 PM

Quote:

Originally Posted by martin (Post 5236)
2. At the voice prompt, dial *010 xxxxxx (*010 is the VoXaLot SIP-Code. See http://www.sipbroker.com/sipbroker/a...iderWhitePages, and xxxxxx is your VoXaLot number)

I think this should be updated as there are now 3 servers (America, Europe, Asutralia)

dial *010 xxxxxx if the number is on the american server
dial *031 xxxxxx if the number is on the european server
dial *061 xxxxxx if the number is on the australian server

It would be easier that there's is only one server (that will dispatch to the 2 other if needed). But is maybe not so easy to implement.

Olivier

martin 02-11-2007 10:06 PM

Quote:

Originally Posted by twen (Post 5370)
I think this should be updated as there are now 3 servers (America, Europe, Asutralia)

dial *010 xxxxxx if the number is on the american server
dial *031 xxxxxx if the number is on the european server
dial *061 xxxxxx if the number is on the australian server

It would be easier that there's is only one server (that will dispatch to the 2 other if needed). But is maybe not so easy to implement.

Olivier

Any of the SIP-Codes will work for any server. *010 (or *061, *031) will work irrespective of what cluster you are on. We have worked quite hard to make this interoperability seamless :)

twen 03-06-2007 11:48 PM

Quote:

Originally Posted by martin (Post 5380)
Any of the SIP-Codes will work for any server. *010 (or *061, *031) will work irrespective of what cluster you are on. We have worked quite hard to make this interoperability seamless :)

Hello Martin,

Thanks very much, this "feature" is really cool :) So we don't have to remember 3 prefixes ! I also note you do a great job with one unique SIP URI independentel from the 3 servers !

I really apprciate your work, and more often when I see some providers ho only think that VoIP is only to call PSTN numbers....

Danny Z 03-08-2007 03:01 PM

Martin
Is there a way to dial into your account from PTSN and then dial from inside your voXalot account using the registered providers, as if you are dialing from an ATA voXalot registered phone?

affinity 03-08-2007 03:21 PM

Not yet, but it is on the agenda and it _may_ not be too far away.

tomblandford 03-08-2007 03:36 PM

If you do introduce this feature, it would be really great if we could register a mobile number (or two) so that when that number was used to dial SIPBroker, we wouldn't have to login to our account at all, but could just simply dial straight out again on voxalot account. That is to say, the called ID logs you in automatically in the same way as voicemail is accessed from a voxalot registered SIP phone. Does that make sense?! Voipbuster already have this feature which is awesome, but has no dial plan, least cost routing etc.

affinity 03-08-2007 03:42 PM

CLIDs (multiple) is already part of the plan for automatic authentication; it may also be a setup that doesn't answer the call, but calls you back via your dial plan.... such feature may end up being a premium feature though. Time will tell.

Kim-Bjoern 04-21-2007 09:37 AM

Tried many access numbers but failed
 
Hi forum.

This is my first post here.

I have family in Bourgas Bulgaria and have tried different access numbers there - both the ones in Bourgas plus in Varna

My Virtual VoxAlot number is 871001

When I dial the numbers in Bulgaria I get "Welcome" plus some music

I have then tried to dial the following exactly as shown

*010 871001
*031 871001
*061 871001

After few second: "duut duut duut"

What am I doing wrong?

Thanks in advance

Kim

nilamber 04-22-2007 03:26 AM

receiving calls
 
Martin,
I have the xlite softphone configured to my voxalot account and pennytel registered as a provider with voxalot. I can make calls out without any problem but if someone uses the sip broker access number with *010 and my voxalot account number, I never receive calls with xlite and the call is either forwarded to voicemail or my call forwarding number. Any thoughts?
Nilamber

martin 04-22-2007 06:59 AM

Both of your problems are probably due to the same thing - firewall.

In order for inbound calls to work, the Voxalot registrar must be able to send inbound messages to your device.

Take a look at your firewall logs and see if you can see anything.

Kim-Bjoern 04-24-2007 03:53 PM

Does nobody have an answer to my above question? Am I doing anything wrong? I am calling the acess numbers from my cellphone and I have tried now 6 access numbers. I therefore conclude that the system must be out of order?

Kim

244751 04-24-2007 06:48 PM

Quote:

Originally Posted by Kim-Bjoern (Post 8144)
Does nobody have an answer to my above question? Am I doing anything wrong? I am calling the acess numbers from my cellphone and I have tried now 6 access numbers. I therefore conclude that the system must be out of order?

Kim

I believe Martin's response above was also referring to you. CHECK YOUR FIREWALL LOG!!!!

nilamber 04-25-2007 01:01 AM

receiving calls and call forwarding
 
Martin,
In my case you were correct about the firewall blocking receiving calls.
Thanks for that.

I have another question regarding call forwarding. There doesn't appear to be any option on the call forwarding setup page to 'call forward when unanswered'
- this means that when I'm online incoming calls are always forwarded unless I go to the voxalot web site and disable each call forwarding entry (and then re-activate before going offline). Have I missed something?

Nilamber

martin 04-25-2007 01:16 AM

Quote:

Originally Posted by nilamber (Post 8158)
Have I missed something?

Nilamber

Hi,

No this is the way is currently works.

Kim-Bjoern 04-26-2007 10:30 PM

Quote:

Originally Posted by 244751 (Post 8149)
I believe Martin's response above was also referring to you. CHECK YOUR FIREWALL LOG!!!!

Hi,

Well - I'm not sure what a firewall would have to do with it. Because I'm not using a softphone myself. It's an ATA (Sipura 2102) and it's the first unit after my router ZyXEL 660R D1 in order to make QoS work. The only firewall I have is within the PC (Windows own firewall) and its log only mentions accesses granted to different programs. I can make calls with the PC shut down

I have two VoXalot accounts - one for me and one for my wife (on the same IP) One of the accounts is registered with the above mentioned ATA - the other one with a Grandstream GXP 2000 SIP phone. The problem regarding incoming calls via access numbers is the same with both accounts. Both accounts setups are alike but one of them cannot access the mailbox (*500) the other one works fine. People may leave a message if phone unanswered and I do get a pulse sound to tell I got messages but cannot retreive them. I use two-tone POTS phone with the Sipura to dial *500 but get a busy tone already after *5

On both accounts I cannot dial through the access numbers. Maybe some parameters in the SIP setups are incorrect but I'm sure it isn't because of firewalls.

Kim

kurun 04-26-2007 11:22 PM

Quote:

Originally Posted by Kim-Bjoern (Post 8294)
I use two-tone POTS phone with the Sipura to dial *500 but get a busy tone already after *5

On both accounts I cannot dial through the access numbers. Maybe some parameters in the SIP setups are incorrect but I'm sure it isn't because of firewalls.

Kim

Can you access the mail box by dialling "500#" ?
I had a similar problem dialling from a Grandstream ATA, but had no problem dialling from a GXP-2000.

I am not that familiar with the Sipura ATA, but it could be that *5.. is an active feature dialling code used by this ATA, and so the full *500 does not get sent out. Voxalot does not react to 0 or 00.

This was the case with the GS HT-496, and after seeing a thread here, I disabled the features in the ATA setup (it allows that), and the problem was corrected. I can now dial *500 to access my mailbox going through the ATA.

Hope this helps



Quote:

Originally Posted by Kim-Bjoern (Post 8047)
Hi forum.

This is my first post here.

I have family in Bourgas Bulgaria and have tried different access numbers there - both the ones in Bourgas plus in Varna

When I dial the numbers in Bulgaria I get "Welcome" plus some music

I have then tried to dial the following exactly as shown

*010 871001
*031 871001
*061 871001

After few second: "duut duut duut"

What am I doing wrong?

Thanks in advance

Kim

Out of curiosity, I also dialled the Burgas Access Number +35956919080, and I do get a "Welcome" message and music. While the music is running, I dialled my Voxalot account *010-xxxxxx, and I got through to my voicemail with no problem.

Perhaps it was not working correctly when you tried, or it timed out ??

KroontjesPen 08-19-2007 08:30 AM

*10 vs *31
 
There is a difference in using *10 or *31 when dialing the Netherlands SIP Broker PSTN access number in Utrecht +31-30-711-0327, sponsor iXcall.

*10 gives a voice message, off very poor quality, telling that there is something wrong with the number (?).
*31 gives the tone to wait for the connection (call tone ?), however I received no ring tone on my ATA or X-Lite softphone with VoxBasic.

martin 08-19-2007 08:37 AM

Make sure you are using *010 and *031 (both will do the same thing) and *not* *10 or *31

Quote:

Originally Posted by KroontjesPen (Post 11836)
There is a difference in using *10 or *31 when dialing the Netherlands SIP Broker PSTN access number in Utrecht +31-30-711-0327, sponsor iXcall.

*10 gives a voice message, off very poor quality, telling that there is something wrong with the number (?).
*31 gives the tone to wait for the connection (call tone ?), however I received no ring tone on my ATA or X-Lite softphone with VoxBasic.


KroontjesPen 08-19-2007 08:59 AM

Mea culpa.
I am so sorry that I make that mistake.
Every thing works fine now.

KroontjesPen 08-19-2007 10:01 AM

ATA rings phone only when X-Lite is active.
 
Is it needed that you have X-Lite active to let your ATA ring the phone.
When I exit that program I received a message that the number is not available when I call it.
If you can received a call with out that program I have to look further.

martin 08-19-2007 10:46 AM

Quote:

Originally Posted by KroontjesPen (Post 11842)
Is it needed that you have X-Lite active to let your ATA ring the phone.
When I exit that program I received a massage that the number is not available when I call it.
If you can received a call with out that program I have to look further.

No. It sounds like you ATA is not properly configured. Voxalot allows multple devices / softphones to register against the one account.

You might need to look into STUN or do some searching on NAT setup for your ATA.
.

KroontjesPen 08-19-2007 01:42 PM

Solved, but I don't no how. Sorry.
 
This problem is also solved only this time I have no clue what did the trick.
What it was not, that was the firewall.
For that purpose I had ticked every checkbox on to log every error or blocking.

After trying a lot off things I cleared the log file for the x-st time and after some refreshes there was suddenly the notice that SIP1, yes voxalot account, was trying to connect.
First there was a 401 error but the second attempt was good.
From that moment I can call true the Netherlands SIP Broker PSTN access number my ATA.

It is not must to help others but I am very pleased. :D

.

madako 11-04-2007 04:28 AM

How can i dial my beither's PSTN phone using ENUM
 
My brother's PSTN phone is registered with ENUM database e.164.org
Now how can i call the number for free.
here is what i have- my ENUM registered regular PSTN phone
-ATA box registered with my voxalot id and password.
I have configured the box and got a dial tone perfectly.
The main question is this, what device or phone i need to use to call the PSTN phone, and why do i have to gt another VSP, so voxalot is not a VSP?
If i ned to have a VSP then how can i set it up because i already added voxalot to the box and have a dial tone.
Please advise me on how to proceed.

affinity 11-04-2007 04:41 AM

You need VoIP equipment at both ends, each needs to be registered to a SIP service [excepting pure IP to IP setups]. The SIP service could be Voxalot at both ends with different accounts. Each end will be the termination point for it's Voxalot account. You set up ENUM to point to the appropriate termination point. Then with the correct dial plans that use ENUM lookups, you will use ENUM to route the call to the other end.

Without ENUM, you can call directly between the Voxalot accounts using *010{voxalotid}. ENUM adds ability to bypass the PSTN when VoIP can be used at both end points.

madako 11-04-2007 07:08 AM

How can i dial my brother's PSTN phone using ENUM
 
Well thanks for your information, but my understanding from the way i read about ENUM is that,
A- You can call a PSTN line even if the callee is not on the internet as long as the line is registered with enum e.164.org database and active
B- To do so you or me the caller must have an active enum acount number.
Once registered, any call from voxalot My questions are:

1. How can I link my PSTN number to my voxalot account so that anyone using ENUM can call me for free? and vice versa.

chatalot 11-04-2007 08:05 AM

Quote:

Originally Posted by madako (Post 13857)
1. How can I link my PSTN number to my voxalot account so that anyone using ENUM can call me for free? and vice versa.

Go to e164.org and sign up for an account and then add the number with your voxalot sip uri as the alternative number. It may take a few days to a week to be verified and become active in the ENUM database. You will be able to test it by using the test dial box in your voxalot account under smartcalls.

emoci 11-04-2007 08:11 AM

Quote:

Originally Posted by madako (Post 13857)
Well thanks for your information, but my understanding from the way i read about ENUM is that,
A- You can call a PSTN line even if the callee is not on the internet as long as the line is registered with enum e.164.org database and active
B- To do so you or me the caller must have an active enum acount number.
Once registered, any call from voxalot My questions are:

1. How can I link my PSTN number to my voxalot account so that anyone using ENUM can call me for free? and vice versa.

All ENum does is link your regular PSTN number to a SIP URI like 123456@voxalot.com

So if you add your number to the e164.org registry and link it to SIP URI, that's all you do.

Now when someone calls your PSTN number, if they are using a VSP that supports ENum lookups (these VSPs will check the ENum lists for any number you dial, if they find a listing, instead of calling the number they'll call 123456@voxalot.com from the example above). Now the call is completely routed over the internet so there is no cost. One downside is that anyone that uses ENum lookups will now only ring you on the VoIP line, the actual PSTN line will no longer ring when an ENum call comes in (there is ways to address this issue if your VSP provider supports it, eg. setting your VoIP line as DND will cause ENum to tell your VSP that although there is an ENum listing, the line is not available, so it should try the regular PSTN line)

Now for you to call ENum numbers for free, all you need is VoXalot account and a dial plan within that account that has ENum lookup set to Yes. To call ENum numbers from your regular PSTN number, call one of the SipBroker Access Numbers (assuming there is a SipBroker Access Number local to you, and you are not charged for local calls), then enter the number you want to call if full International format CountryCode-Number, or *013-Country Code-Number

In summary: ENum can provide free calls between VoIP users. Where PSTN comes into play is that instead of giving out 123456@voxalot.com or 12345@fwd.pulver.com, you give out your regular telephone number which people are more familiar with. If these VoIP users call the number (and their provider is doing ENum lookups) they'll reach you on your VoIP line. Since the call remains completely within the internet it is free.

Hope that clarifies ENum a little bit. So ENum's usefulness is dependent on people registering their numbers and having a VoIP presence online to which to receive ENum calls.

Hope that clarifies things a bit....

madako 11-04-2007 02:45 PM

How can i dial my brother's PSTN phone using ENUM
 
Thank you so much i cannot express how happy iam now.
Although this might not benefit me very well because my brother is in africa with number starting +234806xxxxxxx, seldom he is online because of the cost and termination is only to a local provider which is PSTN there in Nigeria.
If i have a way to change this that will be too god because iam spending so much money calling home.
Once again thanks.

affinity 11-04-2007 02:49 PM

Perhaps you can SMS him (if he has a mobile) or call him for a very short period WITHOUT using ENUM, then he turns on his Internet and you call him through ENUM or directly using his Voxalot ID if that is the account that is set up for him.

flaquadrat 03-19-2008 02:09 PM

SIPBROKER Access number for Austria
 
Hallo!
SIPBROKER Access number for Austria is not active anymore.
The sponsor "sip-fon.com" does not exists:

Here is cut+copy from the Sipbroker list:
###########################
Austria
City PSTN Number Sponsor Notes
+43-720902010 Sip-Fon
###########################

Regards,
Fla

madako 03-19-2008 06:40 PM

How can i dial my brother's PSTN phone using ENUM
 
My brothwer does not have internet.
So you men ENUM cannot work without internet?

DracoFelis 09-06-2008 11:29 PM

Quote:

Originally Posted by madako (Post 15865)
My brothwer does not have internet.
So you men ENUM cannot work without internet?

ENUM depends upon your perspective.

From the perspective of the person who owns the phone number (i.e. the party being called), setting up ENUM is just a way to tell the world that such and so VoIP address (for example, a VoXaLot account) will reach the same person as a specific normal telco phone number call would. So from the perspective of the person BEING CALLED (i.e. who owns the number), ENUM is just giving the world info about the fact that your VoIP will also reach you (in addition to the normal telco phone route). This doesn't do anything for you per se, but it does make it easier for people to call you "for free".

OTOH from the perspective of the person DOING THE CALLING (i.e. the party calling your number), ENUM is a way to use that info to decide if there is a "free" (VoIP only) way to reach the same person, knowing only their normal telco number. If that person's normal calling allows for such free calls, this is a possible way to make the call for free.

Note:

1) VoXaLot support outbound (calling party) ENUM by default. So unless you turn off this feature (and it is configurable in your VoXaLot account), when you try using your VoXaLot account to call someone in ENUM (via one of your "providers"), VoXaLot will detect this fact and (using the ENUM info) redirect that call to the "free" (VoIP to VoIP) route automatically.

2) VoXaLot doesn't do inbound ENUM (making it easier for you to be called by someone who knows your telco number) be default. However, if you want to let people know your telco number(s) can also be reached via VoIP (for example, via your VoXaLot account), you can easily sign up with the e164.org ( Welcome to e164.org ) service (which also has a support forum here on VoXaLot), to register your inbound telco numbers with ENUM.

3) ENUM will only work if/when it is setup for that number (i.e. the called party has published their VoIP route in ENUM), and the caller is using a service (such as VoXaLot) that makes use of the public ENUM service. And since normal telcos don't generally use ENUM (or if/when they do, they often don't provide discounts to the caller for having done so), that means that a person is using a "normal phone" they will only benefit from ENUM if/when they use a third party servicet that supports ENUM. For example, ENUM calls can be completed (for no cost, beyond any telco charges to call the dialin number), via any of the SIP Broker dialin numbers:
SIPBroker - PSTN Numbers

jilljill 03-07-2009 11:07 AM

as a basic user of voxalot

can i use this feature?

i tried to call the local access number

then after the voice tone

i dialed *010XXXXXX

i am getting engaged tone ?


can anyone help me pls

thanks

kurun 03-08-2009 11:26 PM

What you did seems correct for a call using a SIP-Broker access number.

Can you please identify which Access number you called so somebody can test if it is working correctly?

thunder694 05-07-2009 05:47 AM

Forwarding to Cellphone problem.
 
HI Martin,

This is my first time using Voxalot. I have a premium voxalot account because I thought I could forward all the voip call to my USA cellphone. I don't have any hardware but I am using the voxalot forward (call connection) method.

Here is what I did:

I called in one of the usa PSTN phone number and then *010 347562 and that number forward to my cell phone and ring, but when I received Phone call I don't here any sound for incoming phone or outgoing phone, and probably my pstn phone hung up after forwarding to my cell phone.

Setting that I created for voxalot forward method:

Priority:1
Active: yes
Matching Rule:

Match by: inbound number or provider
Time rule: day

Handling Logic:

Connect to: phone number
Phone number: my cellphone number#
provider: VoipStunt

Tell me if I created something wrong or is there anyway I can receive voip call to my cellphone.

Diagram

PSTN------voxalot number-----provider---forward to cell phone.

So far everything works but can't here any audio from both phone.

I used 2 phone to connect to each other. I called from PSTN number and voxalot forward to my cell phone number. but no Audio from both Place.

let me know how can you help me.

Respectfully,

Farid

Corbu' 05-28-2009 06:00 PM

Quote:

Originally Posted by Danny Z (Post 6437)
Martin
Is there a way to dial into your account from PTSN and then dial from inside your voXalot account using the registered providers, as if you are dialing from an ATA voXalot registered phone?

Quote:

Originally Posted by affinity (Post 6445)
Not yet, but it is on the agenda and it _may_ not be too far away.

Has this feature been implemented yet? If yes, how do I make it work?
For many reasons using an ATA phone is a no no for me (many family members sharing different lines and different cell phones), and this would be "The Ultimate" to make Voxalot complete.

mtjones 09-30-2009 11:18 PM

Calling justvoip from access number
 
I assume I have missed something, so please let me know what else I need to do:

I have registered for the basic service and been assigned a 6-digit number.

I have entered my justvoip account details -- username, password, host address (sip.justvoip.com) and it shows as active.

I call a local sipbroker access number, enter *061 and my 6-digit number, and get a message saying that the personal extension is unavailable.

Any advice?


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