Dial the receiver's 7 digits BBPGlobal extension directly from your IP phone, e.g. '1234567' followed by the 'Dial' button.

Note: Some IP phones uses the '#' key as the 'Dial' button. If you are using a softphone, please use the dial button instead of the '#' key.

BBPGlobal is designed to operate anywhere in the world. Therefore the dialing procedure is also the same anywhere in the world. No matter you are in Toronto calling Toronto, or Hong Kong calling Toronto, or Sydney calling Toronto, or Russia calling Toronto, or Jordan calling Toronto, you dial the same number:

For a detailed dialing procedure table, please click here.

Let’s say your BBPGlobal Extension is 1234567 and your family and friends are everywhere in the world. If they have BBPGlobal they can simply call you by dialing 1234567 on their IP phones. Of course this will be a free call. However if they are calling you from normal PSTN phones, they can call our access numbers in their local cities, then press your extension 1234567 to reach you.

For example: for all your friends in Sydney, you would give your number out as

(02) 8244 4868 ext 1234567#

and for your friends in Auckland you would give your number out as

(09) 970 3151 ext 1234567#

There are several ways that you can operate your voice mail box.

• From PSTN: Dial in any one of the BBPGlobal Voicemail number and enter your extension and password.
• From your IP Phone: Dial 77 from your IP phone.
• From the Internet: Listen directly from our website after logging into your account
• From Webmail: Listen from your extension@bbpglobal.com webmail
• From your own Email: Forward to your own email address.
  

There is no time limit a subscriber may retain each BBP Global extension number, except if the extension number has been inactive for 12 months or more. In that event, at our sole option, the company may cancel the account or reallocate it to another subscriber.

You can check your call details through our website http://www.bbpglobal.com and login with your BBPGlobal extension and password.

Yes, but the quality may not be very good. If you want to use BBPGlobal on a dial-up Internet connection, we recommend you to use a softphone with GSM, G729 or G723 codec, such as the X-Pro. G729 or G723 codec compresses your voice signal down to about 6 to 8kbps and consumes much less bandwidth than codecs such as G711, and results in much better voice quality.

If you use the free SJPhone or X-Lite softphone, only use the GSM codec.

Use the best sip server for you. We provide several free sip servers for you to choose:

• sip1.bbpglobal.com:5190
• sip2.bbpglobal.com

Use a hardware ATA adaptor or IP phone, or gateway. These hardware devices usually can provide better voice quality.

Use the right codec. We recommend our users to choose the g729, g723, or GSM codec. These codecs use bandwidth far more efficiently (8kbps, and 6.3kbps respectively) than the g711 codecs (64kbps).

Choose a good quality ISP. Try to avoid those ISPs who run oversubscribed/congested networks as that will affect the voice quality.

Our tests show that if you are using softphone without a headphone or USB IP phone, the opposite side usually hears bad echo. This can be improved significantly by using a headphone or an USB phone. You can purchase these accessories from us. (see details)

BBPGlobal is a VoIP service that uses the SIP protocol. SIP is one of the most popular VoIP standard being used in the industry. Most of the VoIP phones and/or softphones support SIP. BBPGlobal wants to give you freedom to choose what you like. And what is more, your investment is well spent as you can use the same piece of equipment if you decide to change to another service provider. However we are very confident that you will find our service good value for money and enjoyable to use and will stay will us.

If you have any question about choosing SIP hardware, please do not hesitate to ask us and we will provide you with our unbiased recommendations according to your needs.

I have missed a couple of calls because I do not know anyone is ringing me. What is wrong?

Please ensure that the softphone/hardware is not set to DND (Do Not Disturb) Mode.

The voicemail function may not be supported by some hardware. Moreover, if call waiting is enabled, voicemail function might also be affected.

RFC Definition:
21.4.9 408 Request Timeout
The server could not produce a response within a suitable amount of time, for example, if it could not determine the location of the user in time. The client MAY repeat the request without modifications at any later time.

Some possible causes:
- Due to poor Internet/network condition, called parties become unreachable and our sip server cannot receive proper acknowledgement messages.
- Called parties didn’t go off-line properly.

How to resolve:
- Reduce the sip proxy registration time period to 120 to 300 seconds on the sip device (IP phone, adaptor, or softphone). For Xlite, follow:

Menu -> Advanced System Settings -> Sip Settings -> Register Proxy (s): 120

- If your NAT router is connected to more than one sip device, change the listening ports to different port numbers on each sip client. For Xlite, follow:

Menu -> System Settings -> Network -> Listen SIP Port to set port values, e.g. 5070, 5080, 5090, etc for each sip device, respectively.

If there is still problem, also set “Listen RTP Port” to different values, follow: Menu -> System Settings -> Network -> Listen RTP Port.

RFC Definition:
21.4.18 480 Temporarily Unavailable

The callee’s end system was contacted successfully but the cal lee is currently unavailable (for example, is not logged in, logged in but in a state that precludes communication with the callee, or has activated the "do not disturb" feature). The response MAY indicate a better time to call in the Retry-After header field. The user could also be available elsewhere (unbeknownst to this server). The reason
phrase SHOULD indicate a more precise cause as to why the callee is unavailable. This value SHOULD be configurable by the UA. Status 486 (Busy Here) MAY be used to more precisely indicate a particular reason for the call failure.

This status is also returned by a redirect or proxy server that recognizes the user identified by the Request-URI, but does not currently have a valid forwarding location for that user.

How to resolve:
- BBPGlobal forwards all calls to callee’s voice mail if the callee is off-line.

RFC Definition:
21.4.4 403 Forbidden
The server understood the request, but is refusing to fulfill it. Authorization will not help, and the request SHOULD NOT be repeated.

How to resolve:
- Main reason is the caller does not have the right to make the call, or the callee number is restricted. No need to resolve.

RFC Definition:
21.4.5 404 Not Found
The server has definitive information that the user does not exist at the domain specified in the Request-URI. This status is also returned if the domain in the Request-URI does not match any of the domains handled by the recipient of the request.

How to resolve:
- Usually due to incorrect setup. Check configuration.

RFC Definition:
21.4.1 400 Bad Request
The request could not be understood due to malformed syntax. The Reason-Phrase SHOULD identify the syntax problem in more detail, for example, "Missing Call-ID header field".

How to resolve:
- Most likely due to program bugs in the softphone, IP phone, or ATA.

RFC Definition:
21.4.2 401 Unauthorized
The request requires user authentication. This response is issued by UASs and registrars, while 407 (Proxy Authentication Required) is used by proxy servers.

How to resolve:
- Check and reenter username and password, then restart softphone or reboot IP phone, or ATA.