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Unread 12-22-2008, 12:42 AM   #1
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Default 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


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 (
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)
(eg. *18002223333)

-US/Canada (possible via
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
44800 or *01344800
(eg. Dial directly 44800xxxx... or *013-44800xxxx...)

-Germany (possible via
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 :
-Call SIP URI :
>>To call MSN Messenger users:
-Call SIP URI:
(or, and other MSN acct. domains)

>>To Call Yahoo Messenger users
-Call SIP URI:
-The above connectivity is possible thanks to
-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)....

Last edited by emoci; 02-21-2011 at 03:01 AM.
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