We do not send invoices via postal mail. All invoices can be viewed on our website using our self-care interface; including per call details. Invoices are also automatically emailed to your Cititone Unified Messaging email account that can also be checked using the self-care Interface or with an IMAP email client. 

All support plans and calling features are billed in advance. Depending on which day of the month you sign up for service you will either be billed for the remainder of the current month, or the remainder of the current month and the next month. When you sign up for new service, change services, or add services to your account the following method will apply: 

If adding service prior to the 20th day of the month – You will be charged only for the remainder of the current month. 

If adding service after the 20th day of the month – You will be charged for the remainder of the current month, and all of the following month. 

This issue only applies to new or changed services. After your first cycle you will be charged only 1 month at a time on the 27th of the month for the following month. 

All monthly services are billed prorated with the exception of the 911 Cost Recovery Fee. 

You must maintain a positive balance in your Cititone account in order to use outgoing Long Distance and International Calling Services. Free Long Distance minutes included with all calling plans are first deducted from your account and credited back to your account at the end of each billing period. For example if your calling plan has 300 minutes of free North America Calling, charges for your North America long distance calls will be first deducted from your account balance like any long distance call and charges for the first 300 minutes of North American calling will be refunded at the end of the billing cycle. Credit for the Free North America long distance charges will appear on the following months invoice. 

Please Note: Free monthly minutes must be used in the same month and cannot be saved to be used at a later time. 

You can add funds to your Cititone account balance by logging into the Cititone Self-Care interface, selecting add a additional packages, and adding balance to your account. 

Possible reasons why a credit card would fail are incorrect credit card number, incorrect expiration date, incorrect card holder name, incorrect security code, insufficient funds, declined by the credit card company, and because of address verification failure. 

Please make sure that all credit card details are entered EXACTLY as they appear on the card and monthly statement.

Yes, you can. If you change your current rate plan to a higher priced rate plan, the difference in price between your current rate plan and the new higher priced rate plan will be charged. If you are changing to a lower priced rate plan, the change will only occur at the end of your current billing cycle. 

Credit Cards – Any card with MasterCard, and Visa logos: 

PayPal – Payment by PayPal is accepted but has certain limitations. 

Electronic Bill Payment – Bill Payment service is available from most Canadian banks using Internet web banking or Telephone banking. If supported by your bank, simply add us to your payee list. Please note payments take up to 5 working days to be processed. 

Direct Wire/Bank Transfers – If you do not have access to any of the above we will accept direct wire transfershowever we only accept large transfer amounts and there is a $15.00 processing fee for these types of payments. 

We do not accept Western Union, MoneyGram, personal or corporate checks, money orders or any method of payment not listed above. 

PayPal can be used to pay for Cititone services in the same way a credit card can with one limitation: 

Currently all re-occurring services (such as services with monthly fees) cannot be charged on a re-occurring basis with PayPal. This means that you will be able to buy a monthly service; however it cannot automatically renew via PayPal. 

You can always add additional funds to your Cititone balance using PayPal or a Credit Card to ensure you will have enough funds in the balance to pay for any re-occurring services such as monthly rate plans and other features. 

You will also be reminded a few days before billing begins so that you can top-up your Cititone balance to have funds available to renew your services.

Yes. Cititone requires all payments via PayPal to be from verified PayPal accounts. To learn how to verify your account, please see the PayPal website here: http://www.paypal.com. 

Before using your PayPal account with Cititone please be sure that you have set your PayPal preferences to do instant payments instead of eCheck payments. 

The drawbacks of using eCheck with our services is: 

1 – eCheck payments will require 3-4 business days to clear, and can actually take 7-9 business days for locations outside of the Canada. 
2 – Until an eCheck payment clears the funds or services you paid for will not be available in your Cititone account. 
3 – The funds you transfer with an eCheck will not be available to you in your PayPal account nor will we have access to them until they clear. 

You will be explicitly given the option to choose whether you wish to use eCheck or your PayPal balance/instant transfer when making a payment with PayPal so be sure to pay close attention to which one you choose. 

Once you understand the idea of an eCheck please make sure you want to use eCheck as your payment method so as to not incur disruptions to your Cititone services. 

We do not charge a special fee to change your phone number. If you want to change your phone number you should request a new real phone number using the online ordering system, and then after receiving your new number you can cancel your old number at any time. You will be charged the standard setup and monthly fees to purchase a new number. 

Login to the Self-care portal, select the Payment Options menu, and then the Your credit card menu. Delete any cards you want to remove or change, and then add new cards. We do not allow you to modify a saved credit card for security reasons. 

If you choose to cancel all service with Cititone, your balance can be refunded to your credit card. To request a refund please contact us. 

GST, PST, and/or HST are billed at your local Provincial rate as required by Canadian law. The address is determined based on the credit card address and or service address entered for E911.

Most rates at Cititone are listed in 10th’s of a cent; for example $0.056. Your balance as displayed when you login is listed in whole cents, for example: $5.24. When you make a call to a destination which rate is listed as fractions of a cent – that is the actual cost. Our billing system actually keeps your balance with 10th’s of a cent. So while your balance would show $5.24 it might actually be $5.243. 

In short, we do actually charge the rates that are listed. Please remember that rates are excluding any applicable taxes. 

We do not expire or remove accounts that have a positive balance. For more information please see our Terms and Conditions.

Calls that are forwarded using the DID forwarding feature will be billed at rates based on the Call forwarding rate plan listed on our International Rates page.

Example: If you are using DID forwarding to forward all incoming calls to a number outside of North America (ex. Singapore), all calls that are forwarded will be billed at the International rate as listed under the International Call Forwarding Rate plan. 

We only accept Canadian dollars at this time for all transactions. 

All international credit cards should allow you to purchase services in Canadian Dollars and automatically exchange to your local currency on your monthly statement (NOTE: Your card issuer usually charges fees). 

All calls placed to toll free numbers are billed at the rates listed on the International Rates plan page. 

Some International countries use the Country code 1 and are part of the North American Numbering Plan however they are not part of the United States or Canada. Examples of some of these countries are: 

American Samoa 
Anguilla 
Antigua / Barbuda
Bahamas
Barbados 
Bermuda 
British Virgin Islands
Cayman Islands 
Commonwealth of the Northern Mariana Islands (CNMI)
Dominica
Dominican Republic
Grenada 
Guam 
Jamaica 
Montserrat 
St. Kitts & Nevis
St. Lucia 
St. Vincent & Grenadines
Trinidad and Tobago
Turks & Caicos Islands
US Virgin Islands

Calls to these area codes are billed at International Rates. 

All 3-way calls will be billed according to your rate plan and long distance or International calls will be billed according to our International Rates. 

Please make sure that you have already added voicemail to your account. Voicemail is free and can be added or removed at any time you wish. 

Once you have ordered voice mail you can dial *98 from your phone and you will hear a welcome prompt, follow the prompts to configure your voice mailbox. 

There is no default password for your voice mailbox when you first order it. 

To configure your password call from your phone to your voice mailbox (*98), and follow the prompts. You may also set the password by logging into our Selfcare page www.cititone.ca/selfcare under the Update Passwords section. 

If you have a real phone number assigned to your account they will see your real phone number. 

If you have multiple real phone numbers assigned to your account you can choose which number to display by logging into My Account and editing your preferences. 

The default setting is 20 seconds but can be updated using the self-care interface.

Voice mail and FAX to Email messages are automatically stored in your Unified Messaging Inbox for retrieval by phone, by logging into Cititone Self-care portal or by using an IMAP email client. Please visit our Features page for detailed feature configurations. 

You may have up to 100 Mbytes of messages in your voice mail at one time, and each message may be up to 2 minutes long. 

Voice Mail messages are stored in your account as long as you have free space in your email box. You can also have all new Voice Mail messages delivered to you via email, or download them from your account for archiving. Please note that your all email accounts and DID’s will be suspended for 2 months after an account is closed then purged at which time all mail will be lost.

Yes, we do support inbound caller ID. You will receive the incoming caller ID for the user calling you, incoming call as long as your user agent (UA) supports receiving caller IDs. 

We do not support or guarantee the delivery of International caller ID. If you are able to receive International caller ID sometimes and then are suddenly not able to then there is nothing we can do in this scenario.

Yes, we do support outbound caller ID. By default you will send out the standard dedicated caller ID number on your outbound calls. 

We do not support or guarantee the delivery of International caller ID. If you are able to send International caller IDs for some time and then are suddenly not able to then there is nothing we can do in this scenario. 

Each serviceplan can only have one voice mailbox assigned to it at this time. 

Yes you can. Our services allow you to receive faxes and then send them to a configured email address, or you can download them right from your Cititone account. 

If you wish to use your own device or software to both send and receive faxes then the simplest way is to use a hardware UA with a fax machine connected to one of the analog ports. You can then setup your hardware UA to register to our servers and simply connect your fax machine to the line currently registered to our servers. 

If you do not have a hardware UA or wish to receive your faxes on a computer/software you should make sure that your default/preferred codec is set to G.711 or if supported T.38. 

If you are having problems with sending/receiving faxes you can always contact us for further assistance. 

*Please read the detailed information regarding Cititone’s support of IP faxing (FoIP) below: 

Technically we support fax using either the T.38 protocol or transparently with G.711. However, both largely depend on the reliability/stability of your Internet connection. An Internet connection with regular packet loss, high latency (ping) to our servers, and/or jitter (large variations in the latency) will cause problems for faxing. While we do support it, and it does work – we can’t tell you how reliable it will be on your Internet connection without you actually trying it. We have a number of Frequently Asked Questions (FAQs) that address your Internet connection quality, speed, and such: 

And finally the major factor which may determine your experience with faxing will depend on where you are sending the fax. Sometimes some carriers we purchase our termination through will not have networks that are compatible or sufficient for faxing. We suggest that you try testing to send a fax to a USA/Canada number first (to make sure your settings are correct) prior to attempting faxes outside the USA/Canada.

Faxes are received in your integrated Unified Messaging account which has 100MB of space. Faxes will not be deleted by our system unless you cancel your service. 

You can also have all new fax delivered to you via email, or you may download them from your account for archiving. You will need to edit your preferences in order to enable the forwarding of fax messages to your email. 

You can store up to 100MB of fax messages in your account at any given time. 

You can also have all new fax messages delivered to you via email, or you may download them from your account for archiving. You will need to edit your preferences in order to enable the forwarding of fax messages to your email. 

All fax messages received are saved to your account or, if configured, emailed to you in PDF format. 

Faxes cannot be saved on your account in different formats so you will need the Adobe PDF Reader, or PDF capable software, in order to read your fax messages. 

We do not provide any guarantee for International faxing. If you intend to purchase our services for the purpose of placing or receiving International faxes then please note this is not something we provide direct support for. 

If you are able to fax Internationally then this would be a plus to the services which you have purchased. If you are not able to fax Internationally then we are limited in the amount of support we can provide in these types of situations as faxing over VoIP/IP is by default complicated and is worsened by International communications. 

Please be sure to read our FAQ on faxing over IP here so that you understand fully the complications of faxing over an IP connection.

A monthly fee of $2.99 is applied to each Cititone account requiring 911 service. These fees are used to recover costs directly associated with developing, implementing and maintaining 911 service in compliance with CRTC regulations in Canada. 

Cititone provides 911 service across all provinces of Canada. Cititone does not provide 911 or any other type of emergency dialing service in other countries. 

911 service will only be activated on your account if you are located in Canada and purchase a paid service that requires 911 service to be activated. This includes if you purchase a rate plan to place outbound calls, or if you purchase a phone number to receive calls. 

If you are located in Canada and have purchased a rate plan for outbound calling, or a phone number to receive calls we must provide you with 911 service as per CRTC regulations in Canada. 

911 service is typically activated in less than 1 hour if provisioning is successful. You will receive an email once your 911 service has finished provisioning which will indicate if it has been successfully activated. Please note: that until you receive notification via email that your service is successfully activated you will NOT be able to place calls to 911. 

You will not be able to place a call to 911. You will want to contact us by opening a trouble ticket on your account. A Cititone representative will be in contact to discuss the exact reason behind the failure to provision. 

Many things can cause the provisioning of your 911 service to fail; the most common is an address issue. Should your 911 provisioning fail a Cititone representative will be in touch with you to discuss the specific issue. 

Please note: You will not be able to reach 911 until 911 has successfully been provisioned. 

Instead of using a standard telephone line, telephone calls and sent and received over an Internet connection. A VoIP gateway or phone connected to the Internet converts the analog signals into digital format, which are further broken down into smaller chunks called ‘packets’, before sending them over the Internet to their final destination where they are converted back into analog signals. This technology allows the complete integration of both voice and data networks allowing users to take advantage of the features of both VoIP and the PSTN. 

Cititone Business Trunks can be configured for redundancy using failover routing, conditional call forwarding or call forwarding unreachable using the Cititone Self-Care interface. This allows inbound calls to be routed using the PSTN in the event of an Internet or equipment failure. Similarily customer IP PBX and other VoIP equipment can be configured to route outbound calls using the backup PSTN lines in the event of an Internet failure.

An IP phone uses Voice over IP technologies allowing telephone calls to be made over an IP network such as the Internet instead of the ordinary PSTN system. Calls can traverse the Internet, or a private IP Network such as that of a company. The phones use control protocols such as Session Initiation Protocol, Skinny Client Control Protocol or one of various proprietary protocols such as that used by Skype. IP phones can be simple software-based Softphones or purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone. Ordinary PSTN phones are used as IP phones with analog telephony adapters (ATA).

An Analog Telephone Adapter (ATA), also known as the Analog Telephony Adapter, is an electronic device used to enable one or more analog telephones or facsimile machines for Voice over Internet Protocol (VoIP) calls and faxes. An Analog Telephone Adapter basically creates a physical connection by use of telephone and Internet cables between a conventional phone or fax and a computer or an Ethernet gateway. The Analog Telephone Adapter allows the user to immediately take advantage of the lesser costs involved in making calls over the Internet. It also eliminates the need to get rid of existing phones and replace them with IP specialized phones. Moreover, ATA’s are also generally cheaper than specialized digital VoIP phones which are capable of direct USB port or Ethernet gateway connectivity. 

Software VoIP phones use the PC’s sound card, speakers or earphones, and microphone. This hardware works to emulate a telephone, even though this is not what the PC was designed for. 

For better ease-of-use, many companies manufacture USB VoIP phones. These devices give your PC a normal-looking telephone handset or headset 

A PBX (Private Branch Exchange) is a small telephone switch owned by a company or organization. An IP PBX is simply a PBX which supports VoIP (Voice over IP). An IP PBX can also be referred to as a VoIP PBX. 

An IP PBX may support VoIP both internally and externally. Internal VoIP support means that the IP PBX uses VoIP to communicate with each of its connect PBX phones. External VoIP supports means that the IP PBX uses VoIP to route calls to the outside world 

Most IP PBX’s also support older analog or digital PBX phones and also support external connections on the public switched telephone network (PSTN).

Asterisk and other systems based on Asterisk such as freePBX or trixbox are extremely powerful open source IP PBX server software that can be used to meet the needs of almost any enterprises VoIP needs. 

Asterisk SIP trunks are 100% supported by Cititone making setting up Asterisk, FreePBX, Trixbox or other IP PBX’s easy and straight forward. For complete details on setting up Asterisk and FreePBX trunks to use Cititone, please view our ManualsPage or contact support for assistance.

A VoIP Gateway, or Voice over IP Gateway, is a network device which helps to convert voice and fax calls, in real time, between an IP network and Public Switched Telephone Network (PSTN). It is a high performance gateway designed for Voice over IP applications. Typically, a VoIP gateway comes with the ability to support at least two T1/E1 digital channels. Most VoIP gateways feature at least one Ethernet and telephone port. 

The main advantage of VoIP gateway is that it can provide connection with your existing telephone and fax machines through the traditional telephone networks, PBXs, and key systems. This makes the process of making calls over the IP network familiar to VoIP customers. 

The main functions of VoIP gateways include voice and fax compression or decompression, control signaling, call routing, and packetization. VoIP gateways are also power packed with additional features such as interfaces to external controllers like Gatekeepers or Softswitches, network management systems, and billing system. 

H.323 is an ITU standard multimedia conferencing protocol, which includes voice, video, and data conferencing, for use over packet-switched networks. 

H.323 was the first standard for VoIP, but is most commonly used by service providers for backbone links where SIP is more commonly used by end user devices. 

H.323 defines five components of a multimedia network: 

· Terminals
· Multipoint Control Units (MCUs)
· Gateways
· Gatekeeper 
· Border Elements

Terminals are telephone and PC equipment which connect end-users to the H.323 network. 

MCUs are responsible for managing conferences. MCU’s consist of a Multipoint Controller (MC) and an optional Multipoint Processor (MP). The MC manages signaling and the MP manages media mixing and switching. 

Gateways interface the H.323 network with other networks, including PSTN (Public Switched Telephone Network) and other H.323 networks. Gateways consist of a Media Gateway Controller (MGC) and a Media Gateway (MG). The MGC is is responsible for call signaling functions and the MG is responsible for media-related functions. 

Gatekeepers are responsible for admission control and address resolution. Gatekeepers are able to provide advanced services such as normally found in PBX’s. 

Border Elements are positioned between two H.323 networks and assist in call routing and call authorization. 

The H.323 standard provides a foundation for audio, video, and data communications across IP-based networks, including the Internet. H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that do not provide a guaranteed Quality of Service (QoS). These networks dominate today’s corporate desktops and include packet-switched TCP/IP and IPX over Ethernet, Fast Ethernet and Token Ring network technologies. Therefore, the H.323 standards are important building blocks for a broad new range of collaborative, LAN-based applications for multimedia communications. It includes parts of H.225.0 – RAS, Q.931, H.245 RTP/RTCP and audio/video codecs, such as the audio codecs (G.711, G.723.1, G.728, etc.) and video codecs (H.261, H.263) that compress and decompress media streams. 

Media streams are transported on RTP/RTCP. RTP carries the actual media and RTCP carries status and control information. The signalling is transported reliably over TCP. The following protocols deal with signalling: 

RAS manages registration, admission, status.
Q.931 manages call setup and termination. 
H.245 negotiates channel usage and capabilities. 
H.235 security and authentication.

SIP (Session Initiation Protocol) is an IETF standard multimedia conferencing protocol, which includes voice, video, and data conferencing, for use over packet-switched networks. 

SIP (Session Initiation Protocol) is an IETF standard multimedia conferencing protocol, which includes voice, video, and data conferencing, for use over packet-switched networks. 

SIP is an open standard replacement for the ITU’s H.323. 

SIP is described in RFC 3621 – SIP: Session Initiation Protocol. 

SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. 

SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user’s current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols. 

Other RFC’s which document SIP include: 

· RFC 3262 – Reliability of Provisional Responses in the Session Initiation Protocol (SIP)
· RFC 3263 – Session Initiation Protocol (SIP): Locating SIP Servers 
· RFC 3264 – An Offer/Answer Model with the Session Description Protocol (SDP) 
· RFC 3265 – Session Initiation Protocol (SIP)-Specific Event Notification 

A codec (Coder/Decoder) converts analog signals to a digital bitstream, and another identical codec at the far end of the communication converts the digital bitstream back into an analog signal. 

Choosing the best codec for particular network conditions will considerably increase the quality of voice calls. Although G.711 offers very good voice quality over LAN networks, it’s packet size and bandwidth usage make it an unsuitable choice for Internet voice trunks. Much better results when sending calls over WAN links are achieved using the G.729 protocol due to it’s high sound quality and low bandwidth properties. Since G.729 is not an open source protocol, a software license must be obtained for its use making it unavailable with most free soft phones or low quality VoIP devices. Since most VoIP devices allow the codec to be prioritized, G.729 should always be set as first priority for any device sending or receiving calls over the Internet and all other protocols should be disabled if possible to avoid network congestion. 

In contrast to broadcast-type media transmission (RealAudio, for instance), latency in a two-way phone conversation can be very undesirable. 150ms is the specified maximum desired one-way latency to achieve high-quality voice. Voice users will notice round-trip delays that exceed 350ms. Delays beyond 500ms cause calls to be very low quality and cause a walky-talky type conversation.

The Real-time Transport (RTP) Protocol provides end-to-end network transport functions suitable for applications transmitting real-time data such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers. 

Yes. We accept customers Worldwide but we do require confirmation that your account is going to be used by an established business.

To request cancel your Cititone service or account and to request a refund of any unused balance, please contact us. 

If you see calls to destinations you believe you did not call, please contact us. 

Cititone does not support registering with multiple user agents (soft and/or hard phones) at the same time. All Business trunks are to be used by 1 device or PBX at a time to protect user security. 

A user agent is the software or hardware that you use for calling. 

Software user agents (or softphones), allow you to place and receive calls using your computer, a microphone and headphones. There are also other software user agents which function like IP PBX systems, such as Asterisk, 3CX, Linksys…etc. Any SIP/2.0 compatible software is an example of a software user agent. 

Hardware user agents are equipment that connect to your broadband Internet connection (such as DSL or Cable) and let you place and receive calls using a traditional telephone. Hardware user agents may be IP phones, hardware PBX systems or pretty much any SIP/2.0 compatible hardware.

All calls, inbound and outbound, are limited to 120 minutes (One Hundred and Twenty Minutes). The main reason behind this policy is to protect against mis-billing and fraud. 

We are also limited by the industry standard 120 minute/2 hour call limit. Most carriers, both International and domestic, set this limit on their network for similar fraud/billing reasons and as such we also limit calls on our network to the 120 minute standard for interoperability reasons. 

If you are disconnected on a call you may simply place your call again. 

If you have any further questions regarding this limit you may contact us. for further assistance. 

Trunks and channels are both terms used to describe the physical circuit used to make outbound calls or receive inbound calls. The number of trunks or channels each calling plan has indicates the number of simultaneous inbound and outbound calls you can make at one time. Calling plans with 2 channels/trunks can have 2 simultaneous calls while calling plans with 8 channels/trunks can make up to 8 simultaneous calls. The number of channels required should be determined based on your organizations individual communications needs. For more information or assistance please contact us. 

All standard Cititone VoIP Business Plans include free incoming calling with the exception of calling plans using Toll Free DID numbers. Toll Free numbers are charged according to your service contract. For more information please contact us. 

Long distance charges are charged by the minute according to the rate posted on our International Rates Page. 

You must maintain a positive balance in your Cititone account in order to use outgoing Long Distance and International Calling Services. Free Long Distance minutes included with all calling plans are first deducted from your account and credited back to your account at the end of each billing period.

For example if your calling plan has 300 minutes of free North America Calling, charges for your North America long distance calls will be first deducted from your account balance like any long distance calls and charges for the first 300 minutes of North American calling will be refunded at the end of the billing cycle. Credit for the Free North America long distance charges will appear on the following months invoice. 

Please Note: Free monthly minutes must be used in the same month and cannot be saved to be used at a later time. 

You can add funds to your Cititone account balance by logging into the Cititone Self-Care interface, selecting add a additional packages, and adding balance to your account. 

If you have a Canadian phone number assigned to your account it is likely that this number belonged to multiple people before it was assigned to you. 

We do not re-assign numbers that were assigned to another Cititone user within a short period; instead we hold numbers for at least 2 months before assigning it to another user. Unfortunately many telemarketers use out of date databases to conduct sales or research and you may receive calls from them. 

We suggest that you visit the National Do Not Call registry and register your number. 

Please visit: http://www.lnnte-dncl.gc.ca 

Cititone users are able to select the DID they would like from our Sign-Up pages. If you already have a Cititone account and would like to purchase an additional number, you can do so using the Self-care interface by selecting add an additional service and selecting the package that best suits your needs. For Vanity numbers and Toll Free numbers please contact sales .

Yes in some cases. We do support Local Number Portability (LNP) in some areas. 

Currently LNP is only available in some areas of Canada. LNP is not currently available within any area outside Canada. 

We can port your number if you meet the following 3 conditions: 
1. You will need to verify that your number is portable to us using our LNP tool. If it isn’t portable at this time then you may check again in the future to see if we are eventually able to port your number. 
2. We generally ask that customers try our phone number service first to make sure they are happy with the service prior to performing a port. 
3. There is a $75 charge per number for porting. This fee is non-refundable. If you would like to port a sequential range of numbers please contact us for details. 

If you are interested in porting your number we strongly recommend you first try our service and even purchase a phone number to see that it works in the way you expect before doing the port. This is to make sure that you understand how the service works; and that you are happy with it prior to beginning the porting process. Once ported you can also easily switch your current number to your ported number without adding a second product, or you can of course add a second product. 

Because the porting process is time consuming and difficult for everyone (including you) we want you to be completely satisfied with Cititone before porting your number to our service. 

If your number is portable to us please first test out our services and, once you are comfortable with what you experience, place an LNP order right from your account. .

Yes; for Canada phone numbers. We will not prevent the porting of any Canada number from Cititone to another provider. 

While numbers purchased in other countries may be portable to other providers; we do not have information on local regulations and porting procedures that vary from country to country and carrier to carrier. 

The average time to port a number is 25 days. Unfortunately there are no industry standards on porting time to a landline facility (Cititone while based on VoIP is still considered as a landline facility with respect to porting); and generally the “losing” (existing) carrier is the place where ports get stalled for the most time. 

We cannot provide an exact date for when your number will be ported when you place your order to port; but we will notify you (normally at least a few days in advance) of the exact date your number will port. 

To begin porting your number, after reading our LNP FAQ, please follow the instructions below: 

1 – Visit our LNP verification page, if you haven’t already, to check your number 
2 – If your number is portable to us simply click on the “Begin porting” button at the bottom of the page 
3 – Follow the instructions and complete the order form 

*If your number is portable to us please make sure to have a copy of your most recent invoice or bill from your current service provider. Also make sure to sign the Letter of Agency document that you printed out after completing the LNP order form. 

Once you have completed the LNP order form a trouble ticket will be opened on your account and you will be updated at every step of the LNP order process. 

Yes, you may cancel a port order you have placed with Cititone up until we receive a confirmed porting date for your number by updating the trouble ticket that is tracking your port order. 

If we have received a confirmed porting date from your existing carrier already we may not be able to cancel your port order depending on how soon the port is scheduled to occur. 

We cannot guarantee or select a specific date your number will port on. 

Yes. Once you have signed up and logged into the account management portal you will be able to manage all aspects of your account including ordering additional services. 

Contact Information
Name, Address, Telephone #, email addresses 

Login/General information
Username and password for web access 
Username and password for SIP device/software access 
Forgotten password question and answer 
Time zone 

Calling Preferences
Caller ID to send 
Do not disturb 
Caller ID blocking 
Call waiting 
Anonymous call rejection 

Voicemail Preferences
Voice mail timeout 
Voice mail email alerts 

Phonebook/directory
Phone book / Speed dial 

Reports 
Tracking of all inbound and outbound calls and voice mail received 
View transactions 
View bill statements 
View call history 

Products and Services
Rate plans ordering and modifications 
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NO. Our employees will never ask you for any of your Cititone password by phone, email, within trouble tickets, or in any other way. 

DO NOT ever give any of your password to anyone. 

If you have given any of your Cititone passwords to anyone, or you believe your passwords have been compromised please contact support immediately. 

You can safely enter your entire credit card number and other personal information via our secure server, which encrypts all submitted information. 

You can tell the server is secure by lock icon shown in the bottom of your browser window. 

If you are suspicious that the website you are visiting is not the real Cititone website, you simply enter http://www.cititone.ca into your browser bar, instead of clicking on a link within another page or email you have received. 

If you believe your passwords or other personal information have been compromised please contact support immediately. 

Yes. The My Account Management website including the login and all management pages are located on secure servers, and the communication between your computer and our servers is encrypted using SSL. 

If you completed the first step of the sign up process but did not receive an email with the verification link then you should to do the following: 

1 – Check your spam / junk filters and make sure that emails from Cititone are not being deleted. 
2 – Check your trash folder to make sure that the verification was not accidentally, or automatically, deleted. 

If you are still unable to find or receive the verification link then you may easily try signing up again with the same email address. If you still do not receive the verification link you can easily use another email address to sign up for Cititone services. Once signed up you can then add your primary email to your account. 

Please note that until you complete step two (2) of the sign up process, which can only be completed if you received the verification link, your account will not be created. 

If you forgot your password you may follow the instructions below to recover it so that you may log in to the Self-care portal: 

1 – Visit this page and follow the step by step instructions to enter a new password. 
2 – Check your trash folder to make sure that the verification was not accidentally, or automatically, deleted. 

If you are still unable to log in to your account or you are unable to recover your password then you may contact us for further assistance. 

If you forgot your SIP / Phone password, used to configure your UA, then you can follow the instructions below: 

*Note that when you sign up for a Cititone account your My Account Management and SIP / Phone password are the same password you initially entered during the sign up process. 

1 – Login to your account and click on the preferences link 
2 – Change your “Phone password (SIP password):” to the password you would like to use. 

Once you have successfully changed your password edit your UA configuration to match the password you have just created and begin enjoying our services. 

In order to register your UA (User Agent) to the Cititone servers you will need the correct login information. Specifically you will need the following: 

Username/Auth ID: Your assigned number/MYCTID 
Password: The password you created initially. You may also change your password using the Self-care interface. 
Proxy Server: Cititone SIP Address 
Domain/Realm: Cititone SIP Address 
Registrar Server: Cititone SIP Address 
Port: 5060 

For detailed configuration information for most of the common ATA, IP Phones and IP PBX’s, please view our manuals page. 

If you have already changed your SIP/Phone password and are still having problems registering to your Cititone account due to your password not taking effect then please follow the instructions below: 

1. Login to your UA (User Agent), software/hardware, and update it with the new password you wish to use 
2. Once you have done this save your changes and either unplug the hardware or exit the software to completely disable it at this time 
3. Login to your Cititone account and click on the preferences link 
4. Change your “Phone password (SIP password):” to the password you used above 
5. Wait 2 to 3 minutes before starting up your UA again 

At this point your UA should properly register to your Cititone account. 

The reason for this is because login information is cached in the Cititone authentication servers. You need to first make sure that your UA stops attempting to use your old login information, which would cause that information to be updated and may increase the time it takes to change your login information. Keep in mind that this is not the same for all UA however these instructions should work for most scenarios where password changes do not take immediate effect. 

Getting support with Cititone is easy and efficient, saving you both time and money. Our Live Chat support is available 24x7 through our Self-Care portal. Simply open a trouble ticket with the details of your problem and start a chat session with our engineers. 

We only provide support via live chat and through our trouble ticketing system. 

Your user agent (soft or hard phone) does not have G.729 codec’s enabled. Your user agent must support and have G.729 codec’s enabled to receive calls on your real phone number. 

If you are already a Cititone customer, you can test your bandwidth by using our Bandwidth Test application which is available on the Self-Care Interface by following these simple steps. 

. Go to http://www.cititone.ca/selfcare, enter your Username and Password. 
. Click on the Troubleshooting tab and then click Speed Test. Cititone¡¯s system will test your internet speed automatically and display the results along and compare them with our minimum requirements for concurrent calls. 

The results from this test will vary widely depending on factors such as: 

A. How busy the server is when you run the test. 
B. Your Internet Service Providers backbone links. 
C. How far away you are from the server you test with. 
D. And of course your actual connection type. 

If you do test using our application and find that your Internet connection is not delivering the speed you expect; please contact your Internet Service Provider for help or to inquire if they have their own speed test site in order to get another sample. 

The most common problems for not being able to register are: 

1. The password you entered into your user agent is incorrect. Make sure that you are typing your password exactly as you entered it when signing up for a Cititone account. If you do not remember your password then you may easily change it. 
2. If you have setup your user agent with Static IP’s please be sure that you have entered Domain Name System (DNS) server addresses provided by your ISP or your user agent will not be able to register. 
3. Make sure your Internet connection is working, and that your user agent is able to “see” the Internet. 

Most likely you are dialing an International number in the wrong format. The proper rules for dialing through Cititone are as follows: 

US / Canada / Countries located within country code one (1): 1 + area code + number. 

For International calls : 011 + country code + city/area code + number (you may also use 00 instead of 011). 

Poor call quality is usually caused by poor quality Internet connections, improper codec configurations, packet loss on your Internet connection and LAN, or a combination of all of these scenarios. 

If you continue to experience poor call quality, even while calling our test number, there are a few things you can check to make sure that the problem you are experiencing isn’t related to your configuration or connection. 

You can first make sure that you are using G.729 as your primary/default codec or that your codecs are setup according to our recommended order of codecs. 

You can also check to make sure that you are not experiencing packet loss and that your Internet connection has sufficient capacity available by using our looking glass to ping your local IP address or by using our bandwidth test in the Self-care page. 

Please read our how to in order to learn more about resolving quality problems with your Internet connection. 

If none of the above tests can reveal or solve the cause of your poor call quality experience then you may contact us for further assistance. 

We only allow pinging to a specific address within our network. To check the latency from yourself to Cititone’s servers, you can ping lg.cititone.net. 

In Microsoft Windows you can ping by following these steps: 

1. Click the START menu 
2. Select “Run…” 
3. Within the run window enter “command”, without quotes, and click the OK button to get to the DOS prompt. 
4. At the command prompt (C:\>) type ping lg.cititone.net and press the ENTER button. 

You should now see output something similar to this: 

Pinging lg.cititone.net [66.199.129.3] with 32 bytes of data: 

Reply from 66.199.129.3: bytes=32 time=20ms TTL=62 
Reply from 66.199.129.3: bytes=32 time=20ms TTL=62 
Reply from 66.199.129.3: bytes=32 time=20ms TTL=62 
Reply from 66.199.129.3: bytes=32 time=20ms TTL=62 

Ping statistics for 66.199.129.3: 

Packets: Sent = 4, Received = 4, Lost = 0 (0% loss), 
Approximate round trip times in milli-seconds: 
Minimum = 20ms, Maximum = 20ms, Average = 20ms 

*Note that your latency will differ from the 20ms shown above 

In an Apple/Mac/Linux environment you will need to launch a terminal and then run the ping command. The terminal application is usually located in Applications->Utilities->Terminal.app. 

A successful ping command Windows systems usually stop at 4 pings while Linux, Apple/Mac and Unix systems will generally keep going until stopped (Ctrl+C). 

Conversely to ping your computer or Internet router IP from our routers, login to our Self-care site and use our extended ping tool. 

If you, or other users calling you, receive an error message stating that “…the user you are calling is not available…” you will want to check a few things to make sure that you are actually able to receive incoming calls. 

*Note that if you receive the message above this means that your incoming calls are being delivered to your account and that they are failing because of a mis-configuration or other, normally, user related error. 

Since this message can be caused by different scenarios we will point out the most frequent ones: 

1 – If you are doing call forwarding to a PSTN number please make sure that you have a positive balance in your account to forward your calls. 
2 – If you are not doing forwarding to a PSTN number but instead are receiving your calls directly on your UA please first make sure that it is registered to Cititone’s servers. We cannot deliver incoming calls if your UA is not registered to our servers. 
3 – If your UA is registered please make sure that your have configured it properly and that you do not have “Do not disturb” enabled either on your account or in your UA. If you are using Asterisk please make sure that you have configured your inbound routes, trixbox/freePBX or Asterisk, properly. This also applies to other IP PBXs, such as Linksys, which support advanced inbound call routing. 
4 – If you are sure that you have configured all of the above and still do not know why your inbound calls fail please check your account settings to make sure that you do not have “Anonymous call rejection” turned on, that is if you really do not want to receive anonymous calls, or that you do not have “Do Not Disturb” turned on. These can both be activated through star (*) codes so you may have accidentally activated them. 

If you are still unable to receive incoming calls after verifying the above then you may contact us for further assistance.