![]() |
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 : Code:
username_at_gmail.com@gtalk.gtalk2voip.com -Call SIP URI : Code:
username_at_GoogleAppsDomain.com@gtalk.gtalk2voip.com -Call SIP URI: Code:
username_at_hotmail.com@msn.gtalk2voip.com >>To Call Yahoo Messenger users -Call SIP URI: Code:
username_at_yahoo.com@yahoo.gtalk2voip.com -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: Code:
*883975*SkypeUser@sipbroker.com ....(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 |
Quote:
-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 |
|
Quote:
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. |
Quote:
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) |
Quote:
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
|
All times are GMT. The time now is 12:44 PM. |
Powered by vBulletin® Version 3.7.2
Copyright ©2000 - 2024, Jelsoft Enterprises Ltd.