Frequently Asked Questions about VoIP
Finances and General
Voicemail Access Codes:
You can access your voicemail with any device/system connected directly with your account using the codes below:
- *97 (Asterisk 97) is used to access directly the Mailbox associated to the account you are dialing from. If you would like to check which mailbox is associated to your account refer to Assign Voicemail
- *98 (Asterisk 98) is used to access your Voicemail and choose one of your Mailbox accounts. (Will prompt for Mailbox ID and Password)
If for any reason you do not have access to our VoIP network, you can check your Voicemail by just dialing your DID. Once the Voicemail system answer your call, press the asterisk key (*). You will be prompt for Mailbox ID and Password, once logged in to your Voicemail, press 0 (zero) for options. You can also record your greeting and temporary greeting from there.
Once you access your voicemail you're going to be prompted with the number of new and/or old messages you have in the mailbox. Here's the list of options you have with the voicemail system of jimmyTel.com
- 1 - Play the first new/old message available in your mailbox.
- 2 - Change folders. This option allows you to change to another folder in order to hear the messages you have stored in that folder. At the moment it's not possible to change the name of the folders.
0 - New Messages 1 - Old Messages 2 - Work 3 - Family 4 - Friends # - Cancel
- 0 - Voicemail options. In here you can change your greetings and record your name, also you can change the password for your voicemail.
1 - Unavailable message 2 - Busy message 3 - Name. 4 - Temporary message 5 - Change Password * - Return to the Main Menu
Note: At the moment the Unavailable and the Busy message are not working with the voicemail system of jimmyTel.com. You can use the temporary greeting, that overrides all your standard greetings.
The follow options are available when you're listening to your messages. However you can use this options inside the main menu, and they are apply to the first new/old message.
- 3 - Advanced Options
1 - Sent a reply. Currently not available. 2 - Message Envelope. Tell you the Date and time in which you receive the message. * - Return to the Main menu.
- 4 - Play the previous message.
- 5 - Repeat the message.
- 6 - Play the next message.
- 7 - Delete the current message, without confirmation.
- 8 - Forward message to another user. Prompt for the internal extension number.
1 - Prepend the message with a recording. 2 - Send the message without any prepend message. * - Return to the Main Menu
General and Finance Questions
We use e911 (Enhanced 911) and are 100% compliant with FCC and CRTC and cover 100% of USA/Canada.
No. There's no volume commitment and you can spend your credits at the pace you want. Your balance never expires.
We accept PayPal, Debt and Credit Cards.
Minimum deposit is $25USD
At the moment, we have phone numbers (DIDs) available for USA in 49 states and 7 provinces of Canada. We also have Canada and USA toll free numbers. We have international phone number (DIDs) available in more than 30 countries.
Value is the greatest price we could find for Canada and USA. This permit us to offer Canada starting as low as 1/2 cent per minute, depending on location, and USA at a flat rate of 0.012. Our value rate is of the best quality we could find and targeted at end users and resellers who are looking for the best wholesale prices without any volume commitment.
Premium has a flat rate of 0.013 (1.30¢) for both USA and Canada and is routed through established and renowned tier-1 carriers always delivering the same level of quality, at a price that is a little higher than our value option. This price is intended to end-users with critical business calls or resellers who are willing to pay a little more for assured quality, but still less than other US48 tier-1 providers.
Which one to use? Best to do is to try both (value and premium) and settle for the one that best suit your needs.
Yes, you can access your Call Details report online. It is updated every 60 seconds.
Yes, from the customer area, you can download Call Detail Reports (CDR) in CSV, Excel, XML and SQL format. We also have a printable version of the CDR.
If we need to change rates on some destination, including raising or lowering a price, these changes are made on the first week of each month. When you download our rates, there is a column indicating old rate and the date of the change.
USA and Canada: 6 seconds initial, 6 seconds increment
Mexico: 60 seconds initial, 60 seconds increment
World: 6 seconds initial, 6 seconds increment
It's the way we calculate our rates in order to bill your calls. For example, if you call USA for 10 seconds, you will be charged for 12 seconds (2 x 6 seconds since this is a 6 seconds increment call) of a minute, not the whole minute.
We do not accept dialler traffic at this time.
On our premium route, All US/Canada destinations will receive proper caller id. On the value route, we can not guarantee caller id will pass 100% of the time but it should in most cases.
For your convenience, we support 3 standards.
011 Prefix, 00 Prefix and direct country code. For example, to call to UK you can use 01144+number, 0044+number or 44+number. To call to USA you can use 1+Area Code+Number or 001+Area code+number.
Yes we do provide routing (termination) in every country. We use an A-Z VoIP call routing (termination) provider. We always do our best to find and keep quality working routes. If you were to experiment some problems with a particular destination, let us know and we'll make everything possible to fix the problem.
G.711 (μ-law / pcmu) , G.729 and GSM.
For best sound quality, if you have the bandwidth available, we recommend G.711u. However, you can still maintain an excellent voice quality and lower bandwidth usage with codecs like G.729.
We support SIP and IAX2.
"The Session Initiation Protocol SIP) is an application-layer control (signalling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences." (cit. RFC 3261). It was originally designed by Henning Schulzrinne (Columbia University) and Mark Handley (UCL) starting in 1996. The latest version of the specification is RFC 3261 from the IETFSIP Working Group. In November 2000, SIP was accepted as a 3GPP signalling protocol and permanent element of the IMS architecture. It is widely used as a signalling protocol for Voice over IP, along with H.323 and others.
Source and more information: Wikipedia
IAX is the Inter-Asterisk eXchange protocol used by Asterisk, a dual licensed open source and commercial PBX server from Digium and other softswitches and PBXs. It is used to enable VoIP connections between servers, and between servers and clients that also use the IAX protocol.
IAX now most commonly refers to IAX2, the second version of the IAX protocol. The original IAX protocol has been deprecated almost universally in favor of IAX2
Source and more information: Wikipedia
Source and more information: Wikipedia
We recommend SIP. However we fully support IAX2
Any type of software or device (unlocked) which support SIP or IAX2. Including free softphones such as X-lite, free open source PBX Asterisk, Voxalot, Trixbox distribution, VoIP ata's and VoIP phones, Hardware CISCO VoIP switches etc. Basically, any hardware or software that support one of the 2 protocols we offer.
We have VoIP servers located in:
- Atlanta, GA (atlanta.jimmyTel.com)
- Chicago, IL (chicago.jimmyTel.com)
- Dallas, TX (dallas.jimmyTel.com)
- Houston, TX (houston.jimmyTel.com)
- Los Angeles, CA (losangeles.jimmyTel.com)
- New York, NY (newyork.jimmyTel.com)
- Seattle, WA (seattle.jimmyTel.com)
- Tampa, FL (tampa.jimmyTel.com)
- Montreal, QC (montreal.jimmyTel.com)
- Montreal 2, QC (montreal2.jimmyTel.com)
- Toronto, ON (toronto.jimmyTel.com)
- Toronto 2, ON (toronto2.jimmyTel.com)
- London, UK (london.jimmyTel.com)
you may use the server of your choice at any time.
Usually in order to get better results, you should choose the server closest to your location. You can still send a ping to any of the servers to check the best response time.
Our VoIP servers consist mainly of OpenSER and Asterisk running on Redhat Enterprise 64 bits OS.
English and French mainly by email.
Our live support hours by phone are from 9am to 5pm EST (Eastern standard time). However, we do have staff attending your email requests beyond these support hours. Check at the bottom of the page to check current EST time.
We always do our possible to have you up and running. Customer support is an important part of our philosophy. We'll do the best to help you no matter the type of equipment you are using. In the case of Asterisk, we can have a technician enter your server and configure the basics for you (Working trunk and extension).
We do have configuration samples for Asterisk, FreePBX/Trixbox, Voxalot, Generic LinkSys/Sipura ata/phone. We'll add more configuration examples in the future. Configuration to use our service is very straightforward with most softwares/devices. If your equipment is not included in the configuration examples and you have difficulties setting up jimmyTel.com, we'll be glad to assist you.
Yes we do. We've gone as far as programming call shop interfaces and setting up multiple load-balanced asterisk servers for call centers. Contact us.
FAQ Revision 1.42.