The SipBroker Gateway
 

Given that the role SipBroker plays in bringing many VoIP setups together is overlooked, I though I'd put together a quick review of what you can do with SipBroker

1. Accessing the SipBroker Gateway

-Call any of the SipBroker Access Numbers


or


Note: The access methods below using third party access numbers are not officially supported and may change or be discountinued at any time.


-Call any of Bezecom Access Numbers then enter 538802

-Call any of the iNum Access Numbers then enter 883-510-074-022-302

-Call any of the Point One Access Numbers, select 1, then enter 1-747-402-2302 ( may not work due to Gizmo blocking VoXalot )


-If no Access Number is available in your area:
>Access GizmoCall, login with your own Gizmo5 acct. You can now call any SipBroker destination by simply entering *SipCode-Number (you'll need a headset and microphone).
>Access SipCodeBrasil FlashPhone, enter the *SipCode-Number combination you want to reach (you'll need a headset and microphone).

-VoXalot and Gizmo5 users can directly call any SipBroker destinations from their SoftPhone or ATA.

-Users of providers without peering codes can integrate SipBroker dialing into their dialplans


2. What can you do once you've accessed the SipBroker Gateway

>>Call users of over 2000 VoIP networks:
-Locate the SipCode for the Network you're trying to reach (format is *123 or *1234)
-Make sure you know the internal number of the person you're trying to reach
-Dial *SipCode-Number (eg. *010-123456 for a VoXalot user, and *747-17472223333 for a Gizmo5 user)

>>Call any numbers for which an eNum record exists:
-Check that the number you're trying to reach has an eNum record (enter CountryCode+Number)
-Dial number when prompted by gateway directly and talk for free (in most cases the number is to be dialed in international format, eg. 14162223333 or 44-20-7099xxxx or 39-06-91650xxxx etc. )
-Dialing directly or adding *013 to the front (eg. *01314162223333) is equivalent.

>>Call iNum Numbers (INum.net):
Option 1: Dial 883 XXX XXX XXXXXX directly (example 883-510-000-000-091 for the INum echo test number) when prompted by the gateway. You can add *013 to the front to achieve the same result. This is possible due to INum adding ENum records for all their numbers (see iNum – One number for the world » Blog Archive » ENUM for INUM, iNum – One number for the world » Blog Archive » ENUM for iNum Update )
Option 2: Dial: *883946*-INum Number (Possible via SipCode Brazil)

>>Call Toll Free Numbers:

-US/Canada(sponsored by SipBroker)
*1800
*1866
*1877
*1888
(eg. *18002223333)

-US/Canada (possible via e164.org)
1800 or *0131800
1866 or *0131866
1877 or *0131877
1888 or *0131888
(this is different routing than using *18xx above)
(eg. Dial directly 18002223333 or *013-18002223333)

-UK (possible via e164.org)
44800 or *01344800
(eg. Dial directly 44800xxxx... or *013-44800xxxx...)

-Germany (possible via e164.org)
49800 or *01349800
(eg. Dial directly 49800xxxx... or *013-49800xxxx...)

>>Call Echo Testing Services
-*010*600 (VoXalot Echo Test)
-*393613 (FWD Echo Test)
-*266-301 (Blueface Echo Test)
-*850-301 (IdeaSip Echo Test)
-*747-1-747-474-ECHO (Gizmo5/SipPhone Echo Test)

>>Access Various Conferencing services:
-*747-1-222-XXX-XXXX (SipPhone's Party Line-Choose any 7 digit number to create your room)
-*850-100-XXX-XXXX (IdeaSips Conference rooms-Choose any 7 digit number to create your room)
-*9876-7XXXX (DarkVoIP's conference rooms-Choose any 4 digit number to create your room. Use # to bypass PIN needed)
-*747-1-747-555-2663 ( It lets you access Conference Calling Rooms from Free Conference Call. Unclear if you can still use the recording features from here. For a Gizmo specific conference set one up at Free Conference Call-Gizmo , then dial access number and enter your assigned access code)


3.Other SIP Connectivity pointers

>>To call Google Talk Messenger users:
-Call SIP URI : or
-Call SIP URI : >>To call MSN Messenger users:
-Call SIP URI: (or live.ca, live.com and other MSN acct. domains)

>>To Call Yahoo Messenger users
-Call SIP URI: Notes:
-The above connectivity is possible thanks to GTalk2Voip.com
-The "@" in the original email adress of the user is replaced with "_at_" in the SIP URI call
-On the first attempt the user you are calling may be prompted to accept a new friend related to with an adress related to GTalk2Voip ... from there on incoming calls will appear to come from this user.


>>To Call Skype Users (Possible via SipCode Brazil)
Call Sip URI:

....(more info to be added shortly)....

Hi,

Does anyone use iNum by voxbone?

Tpad are looking into trialling it, would any of you use it?

Steven
Tpad.com

The good news with iNum is:

-Once TPad users are assigned them (as Gizmo and soon VoXalot users will be), these users will be able to call each other among networks without having to worry about remembering SIPCodes etc. (probably more benefitial for users doing TPad >> Other networks since they no longer have to worry about SipBroker integration). So think of iNum as instant peering with all the other networks that are supporting it.

-If VoXBone actually manages to get Telcos to terminate to 883 from landlines, that will be a major plus

On that note:

http://forum.voxalot.com/voxalot-sup...nectivity.html

Although not all the iNum Access numbers may be fully activated, the ones I have tested (some in Canada, and some in Europe), work very well, with excellent quality.

iNum was quite responsive to fix an issue with a number when contacted.

As users become familiar with iNum, I would expect that more people will use the service.

Dutch gateway Budgetphone : new server address
 

was/is: budgetphone.nl
will be: sip1.budgetphone.nl permanent after 15 april

in some cases peering will now fail.

Well it seems the switch-over is done step-by-step,
and each user has another "deadline", so it will be a slow process,
this will also be the explanation, *326 is still in use, but users on the new Budgetphone server, will have no SipBroker service anymore, until the last old server user has been switched over.

Peering is okay now, for Budgetphone via SipBroker, (*326)
(for people on the new Budgetphone server)

When Tpad start iNum?
How I can apply the iNum from Tpad?
I wish to use it .

is there anyway to access bestip network through SIPbroker ? bestip is the only voip not blocked in UAE, AwalFon | BESTip ATA | Internet Telephony Service Provider | PC to Phone | IP phone provider | VoIP phone service provider | IP phone | Broadband telephone | VoIP services | Free VoIP | Voice over IP service | Business VoIP | International rate VoIP

fwd is long ago not working anymore via sipbroker, I tried it so often bu always strait fast busy tone.

Hi, please, is there any possibility, how to reach SipBroker gateway through the Skype? For example, add a new Skype user, which function will be "skype gateway to sipbroker", and via tone-dialing enter the user number, than will be connect. Is there any solution...?

Accessing the iNum.net gateway
 

Hallo, please , at the start of the topic, there are descriptions of some methods, how to reach Sipbroker gateway. And I have a question, is it possible to reach iNum.net gateway as SIP URI ( for example "xxxxx@inum.net" or "xxxxx@sipbroker.com" )? Thanks

It's starting to get frustrating, when I dial using PSTN access, I get the announcement but just a fast busy when I dial Gizmo #. I have tried it straight 1-747 and *7471-747 method. Even when I dial using eziDial from Sipbroker.com my hpone rings and then busy tone. Can anyone please shed some light on this. Nothing changed no can call me and others (friends and family) using PSTN access numbers. I've tried New York, Vancouver, Abbotsford, Surrey.... Please help

There seems to be an issue with Gizmo numbers...

The suggestion has been made that Gizmo may have blocked VoXalot and SipBroker IPs for the moment....

It is not clear why just yet and it has been rather hard getting in touch with anyone at Gizmo
See: Gizmo5 - voip - Make free internet calls from your mobile phone and computer and http://forum.voxalot.com/voxalot-sup...-register.html

There is currently no reliable access to do either at the moment...

In the case of Skype...although I have tested access using Net2Max + VoXeo and the SipBroker gateway is accessible....DTMF tones for dialing are not passed...

The inum.net gateway should be reachable as a SIP-URI however the information is currently not publically available from VoXBone...

Skype IN - Skype OUT
 

Hi,

I have tested SIP access to Net2MAX PABX --> VoXeo and the PBXes. It works with DTMF.

I call SIP URI from Voxalot Speed Dial. Use Web Phone (899050xxx) or Net2Max 1CC number.
You can use IP adress too. I use it from my ATA, D-Link DVG-1402S.
Important in Net2MAX Web PABX is Voice Greeting Sequence: dial 8990001

Define Short Cuts in Web Dialer for SkypeUser (899099SkypeUser) or dial SkypePhoneNumber (899099xxxxxxx).

I concurr....

what I meant before was:

Skype>>Skype acct. forwarded to VoXeo Number>>VoXeo App sending call to a SIP URI (in this case leading to the SipBroker gateway).

The SipBroker gateway answers and the greeting can be heard...however DTMF tones do not make it from Skype to SipBroker...

I have tested Skype call to Voxeo and Tropo. Voxeo/Tropo Application transfer call to SipBroker gateway Landline number (tested numbers: Montreal 15149089707, Ny York 12124611066). It works with DTMF.

You can also reach your Voxeo/Tropo applications via SIP URI, Skype, and INum.

Emoci, on Sipcode Brasil you will be able to set a Sipbroker's Code for Skype, Gizmo, Inum and clones Betamax. SipCode Brasil - Home

There is permament error when trying to rich sipnet numbers via sipbroker pstn gateways.

SIP-Code SIP Proxy Provider name
*419 sipnet.ru SIPNET

What I can confirm so far...

A Speed Dial for SIP URI 7555755@sipnet.ru withing VoXalot does not work...a similar speed dial to 7555755@sipnet.ru from within CallCentric works fine....

This means there is an issue between VoXalot (and by virtue SipBroker) and Sipnet.ru ...

There is a chance that they could be blocking us...(at the moment Gizmo, LiberaIlVoip and SipGate do for example), but given the .ru TLD there is also a chance they've ended on a blocklist on the VoXalot end...

It could also be that the whole issue is strictly based on SIP interconnectivity and there is no IP blocking of any sort...

I'll bring it up to Martin and/or Craig and see what happens...

No. I suspect a DNS-related issue on voxalot side - NAPTR, SRV, etc. I could be wrong, but...

Emoci, with SipCode Brazil, you can have a Sipcode redirecting to Skype as well

So this is what I understand from that page:

1. You can call the Skype Echo Test by entering *883936426.... but let's say my skype username was emoci....how would someone call me from their system?

2. You can call user sipcodebrasil on skype and then enter a SipCode number....it seems to be limited to SipCode numbers though... I tried *010500(VoXalot's Voicemail) and did not get anywhere...

To be fair I tried to setup an account for testing ... but since they ask for a "Razao Social" I could not get any further...

By accessing SipCode Brazil - Home you can see all the Subcodes to access Skype, Gizmo, Inum and Betamax Clones etc...

Emoci, just send an email to me jc@sipcode.com.br and I will send you a full account with pleasure. Alternatively, you can reach your Skype by dialling *883975*emoci

Use code *799 ob2.tario.ru instead. Same as sipnet.ru, but working currently.

Because someone i know will use sipbroker to call some sip:uri
I tried some sip Acces numbers.
Then i mentioned that a lot of numbers where out of service or not usable.
All these Netherlands PSTN Number are out of service.
Amsterdam +31-20-8005070 DIDWW 2 concurrent calls supported
Hague +31-70-8911177 DIDWW 2 concurrent calls supported
Haarlem +31-23-8009990 DIDWW 2 concurrent calls supported
Rotterdam +31-10-8009420 DIDWW 2 concurrent calls supported
Utrecht +31-30-8903260 DIDWW 2 concurrent calls supported
Utrecht +31-30-711-0327 iXcall Wait for second tone.

And when i call de Sipbroker Inum (ber) 00-883-510-074-022-302 it don't respond on the ivr input. Tested with two differtent phone's.

I have done some updates to the OP, refreshing some of the info and adding some new tidbits...