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| A collection of frequently asked question prior to signing up or the initial questions that our users share In most cases yes, especially if you want to take full advantage of its functionality. While the speed of the Internet connection isn't critical, the key factors that affect call quality are reliability and some available bandwidth. Make sure you choose an ISP with a guaranteed up-time to ensure a consistent and reliable connection. Alternatively if you will be using the system mainly on your mobile device, a 3G connection is sufficient. 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. Call records are stored and are made available to you on a secure basis. Statistics such as source and destination numbers, date and time of each call as well as the duration of the calls are available. You are able to filter call information according to pre-defined criteria, and you may also able to export this information to an Excel spreadsheet. Please note that only records for last two months will be shown. You can also: 1. Use search criteria to filter your search of call records. 2. Export the call records to an excel file. 3. Remove call records from the list. The advent of Direct Inward Dialling (DID) services provides people with access to local telephone numbers in most countries on a worldwide basis. These numbers are then forwarded by companies such as DID World Wide to PSTN (Plain Old Telephone System), VoIP or VoIM devices anywhere in the world. Essentially, this means that:
Over and above these basic functions of DIDs, there are many additional uses for local numbers. For example, consider the issue of privacy. If users place classified advertisements in a newspaper and they wish prospective buyers to contact them without publishing their own, private telephone number, a temporary local number could be provided by the newspaper. For more FAQs on DIDs and numbers, please view the "Phone numbers" category... The services provided by handility are suitable for both residential and business use, but may not be utilised for auto-dialling, continuous or extensive call forwarding, telemarketing, fax or voicemail broadcasting or voicemail blasting purposes. https://handility.com/is-it-for-me
There is a $4.95 signup fee for creating the account. Since calls can be placed the moment you sign up, in order to forward calls to PSTN we recommend a $10 fully refundable deposit. If you decide not to keep your account within the first 30 days, we will happily refund your monthly fee as well as the signup cost, minus any calls you have made. Some select additional features supported by us include:
If there is a feature you would love to see, just reach out to us... Yes. Calls may be forwarded by our infrastructure to:
The forwarding destination is selected by the customer on a per-DID basis via our easy-to-use web interface. We provide support using chat, email, ticketing system and phone. For more info go to https://handility.com/about/get-in-touch-with-us No. There is no maximum limit on call duration for the DIDs purchased from Handility. Yes.The Handility system provides DID numbers that can be forward to URI, VoIP, CUSTOM, GTalk, or PSTN service. We do provide recording IVR (Interactive Voice Response) services. We have professional voice artists whic have previously done recording for companies like Bank of America and Chase. We also provide different language speakers and accents (British English, Australia) Please contact us for more details Interactive Voice Response (IVR) is an automated telephony system that interacts with callers and routes calls to the appropriate recipient. An IVR system (IVRS) accepts a combination of voice telephone input and touch-tone keypad selection and provides appropriate responses in the form of
The number of channels required largely depends on the type and size of your business.Larger quantities of additional channels are usually considered by business customers, call centres, calling card service providers. Yes.This is a standard use of DIDs and is also referred to as PSTN (regular phone) forwarding. This function allows you to receive calls made to your DID numbers on regular land line or mobile phones. No Internet, PC or VoIP services are required and no minutes are charged. You can map all or any of your DIDs to the same PSTN number. Please note that the number must be entered in the following format: country-code+city-code+number. No.Unfortunately, Handility provides DID numbers for inbound and outbound calls only. Yes.Handility supports VoIP protocols used by Asterisk, including IAX. For more information, pleas eview the VOIP category |
| About types of phone numbers, forwarding, porting and other DID FAQs In historical terms, a DID is a feature offered by telephone companies in order to serve the needs of their customers via PBX (Private Branch Exchange) systems. In practical terms, a DID number (also called a DDI in Europe and sometimes referred to as a virtual number) is a local telephone number in a selected country or city. These numbers are then forwarded by DID World Wide to PSTN (Plain Old Telephone System – a regular telephone), Voice over IP (VoIP), SIP, H.323, IAX, or GTalk anywhere in the world. Typically a DID is used to provide local telephone numbers for customers on a worldwide basis, or to provide access from regular telephones to non-PSTN destinations such as VoIP.When dialling an local number, for example an Australian DID number within Australia, please skip the Australian country code “61” and add “0” (zero) at the beginning of the number instead. For example, dial 03 4550xxxx instead of dialling +61 3 4550xxxx. Some VOIP agents and apps require a full country code + area code + number format. Phone.Systems app can be configured to skip this and enable local dialling. The term virtual number is used because a DID may be thought of as an “alias” for your primary phone number or VoIP destination. A virtual telephone number lets you choose a telephone number from other local calling areas outside of the geographical location where you answer your telephone. You can have as many “aliases” as you wish in cities all over the world. All calls to these virtual numbers will ring your main telephone line or VoIP device as if the subscriber dialled your primary number. For example, you live in Toronto, Canada and you purchase DIDs in Paris, London, New York and Shanghai. Then you configure those DIDs so that all calls made to those virtual numbers are forwarded to your telephone in Toronto. Essentially, virtual numbers allow you to add telephone numbers to your phone line so that you are no longer restricted to one telephone number per line. Each Toll-free number provided by Handility is equipped with 2 dedicated channels at no extra cost to the user and can be increased upon request.
Yes. Porting in/out of phone numbers to Handility is supported. Porting fee may be applied. For more details please contact our LNP Department by e-mail This email address is being protected from spambots. You need JavaScript enabled to view it. Call forwarding destination of DID may be changed by you any time via our web services. There are no additional charges involved for making these changes. However, if a number previously forwarded to a non-PSTN device (such as VOIP etc) now should be forwarded to PSTN (landline/mobile phone number), then an additional flat monthly charge is applicable. Also, in case you wish to change your PSTN number, please note that country should be supported by the same Zone. In case the country is included to different Zone, new PSTN plan should be ordered. Handility offers advanced Caller ID features.The system gives you a possibility to modify the way your Caller ID (CLI/ANI) will be delivered to the receiver, when using such mapping types as Custom or URI. Once you are configuring your number, you will be able to choose from the following three options:
Additionally, CLI Prefix tab lets you adjunct your CLI with an optional '+' and up to 6 characters, including digits and a '#' sign. Statutory LimitationsPlease note that some countries have strict restrictions on changes to the CLI. You will be notified about the existence of any such limitations before purchasing a DID from a relevant country. * Please also note that while we do our best to correctly identify the CLI for all local calls, i.e. when the call is originated from the same country that the DID number belongs to, international CLI often tend to be hard to control due to the various factors and influences outside the reach of our system. No. Our DID numbers do not support collect calls. National DIDs are also referred to by us as tollfree and shared cost. National DIDs are similar to the local DIDs in their use. The difference is that the price of the call to national DID is a little bit higher than call to local DID. Although, the price is much cheaper than the price of international call. Furthermore, it is not related to one location inside the country, you can make calls from different places inside the country and pay less than even for long distance calls in the same country. Yes.Porting out is supported by Handility.com. For more details regarding number portability, please contact our LNP department. |
| FAQs about 1800 and 1300 numbers and other tollfree, national or shared cost numbers Toll-free number (sometimes referred to as a Freephone, Free call, 800 number) is a special free phone number that is free of charge for the calling party. It allows a caller to reach the destination without being charged for the call. The incoming calls on your Toll-free number can be instantly forwarded to one or more of your business / personal telephone lines or other compatible VoIP / SIP devices of your choice. A shared-cost number (for example an Australia 1300 number) is a solution where the initial cost is covered by the caller (as a nominal fixed fee) but the ongoing cost of a call is handled by the recipient.
Each Toll-free number provided by Handility is equipped with 10 dedicated channels at no extra cost to the user and can be increased upon request.
Additional channels increase the number of simultaneous inbound calls that may be handled with your Toll-free number. Additional dedicated channels in bulks of 10 may be assigned to your Toll-free number free of charge upon your request. For that, please contact your account manager or our customer service by e-mail This email address is being protected from spambots. You need JavaScript enabled to view it.
Yes. Porting in/out of Toll-free numbers to Handility is supported. Porting fee may be applied. For more details please contact our LNP Department by e-mail This email address is being protected from spambots. You need JavaScript enabled to view it. No, unless this charge is included in your pricelist. You may need to pay a per minute fee if your tollfree number is forwarded to a PSTN (normal per minute charges apply)
Toll-free number service is offered on prepaid terms. As a Handitliy user, we recommend you keep a positive balance on prepaid account to acquire and use Toll-free numbers (as well as regular phone numbers, per minute charges or other added services you may wish to add). You do have total control over your cost (by viewing your account page on handility.com and only pay for what you use) Each Toll-free number is billed a fixed monthly fee along with per-minute charges paid by the number owner instead of the calling party.Additional fees may be applied for calling to Toll-free number from Mobile phones in United Kingdom (UK). The rates depend on the local operator. Some fees may change over time due to our infrastructure provider fees changing, however any changes are sporadic and not significant. You can see the monthly statement for your Toll-free number use in your monthly invoice. A monthly invoice can be found in the Billing section of your Handility User Web Control Panel.
Yes.The great thing about tollfree numbers is that they are free for the caller. Therefore, the receiving party is charged per minute. No. You will not be able to receive calls; therefore, please make sure your prepaid balance is positive. If this particular feature is a problem for you, your business or operation, please contact us at This email address is being protected from spambots. You need JavaScript enabled to view it. and we may be able to set-up a solution that suits your needs |
| What is a channel, how many channels do I get per line, what is the difference between a fixed and a shared channel... Channels define the number of simultaneous inbound calls you can receive on your numbers.By default each number provided by Handility is equipped with 2 voice channels; however, you are welcome to increase the inbound capacity of your DIDs by ordering additional dedicated (non-shared) or flexible (shared) channels. Dedicated (non-shared) channels are allocated to a specific DID, and allow additional simultaneous calls on this particular single number. Flexible (shared) channels cover a group of selected DIDs, and allow additional call capacity on all of these numbers, notwithstanding their different area or country codes, e.g. DIDs from Canada, Spain and Australia may well be included in the same group (pool). DIDs within this group will be dynamically allocated with necessary additional channels on a first-come, first-served basis. Please note: this is the case for all mapping options offered by Handility except from call forwarding to PSTN (flat rate and Pay As You Go) devices, since a PSTN line can only work with one channel. By default each number provided by the Handility system is equipped with 2 voice channels at no extra cost to the user.To increase the inbound capacity of your DID numbers you are welcome to order additional dedicated (non-shared) or flexible (shared) channels. For more information about additional channels, please see What is the Difference between Flexible (Shared) and Dedicated (Non-shared) Channels? By default each number provided by Handility is equipped with 2 voice channels; however, you are welcome to increase the inbound capacity of your DIDs by ordering additional dedicated (non-shared) or flexible (shared) channels.Dedicated (non-shared) channels are allocated to a specific DID, and allow additional simultaneous calls on this particular single number. Flexible (shared) channels cover a group of selected DIDs, and allow additional call capacity on all of these numbers, notwithstanding their different area or country codes, e.g. DIDs from Canada, Spain and Australia may well be included in the same group (pool). DIDs within this group will be dynamically allocated with necessary additional channels on a first-come, first-served basis. To view the list of countries, where additional channels are supported, please see What Countries Can I Order Additional Channels For? for special requests please contact us. Please note: this is the case for all mapping options offered by Handility except from call forwarding to PSTN Flat Rate plan, since a PSTN Flat Rate plan can only work with one channel. To increase the inbound capacity of your DID numbers you are welcome to order additional dedicated (non-shared) or flexible (shared) channels.Additional channels boost the number of simultaneous calls that you can receive on each DID. Dedicated (non-shared) channels are allocated to a specific DID, and allow additional simultaneous calls on this particular single number. Flexible (shared) channels cover a group of selected DIDs, and allow additional call capacity on all of these numbers, notwithstanding their different area or country codes, e.g. DIDs from Canada, Spain and Australia may well be included in the same group (pool). DIDs within this group will be dynamically allocated with necessary additional channels on a first-come, first-served basis. Please note: this is the case for all mapping options offered by Handility except from call forwarding to PSTN devices (flat rate and Pay As You Go), since a PSTN line can only work with one channel. To increase the inbound capacity of your DID numbers you are welcome to order additional dedicated (non-shared) or flexible (shared) channels in all of the following countries:
If the country that you would like to purchase additional channels in is not included in the list, please send your request to us. Essentially yes.Shared (Flexible) channels span a selected group of DIDs, and allow additional call capacity on those numbers. Flexible (shared) channels cover a group of selected DIDs, and allow additional call capacity on all of these numbers, notwithstanding their different area or country codes, e.g. DIDs from Canada, Spain and Australia may well be included in the same group (pool). DIDs within this group will be dynamically allocated with necessary additional channels on a first-come, first-served basis. Please also note that by default each number provided by Handility is equipped with 2 voice channels; therefore, additional channels are necessary only if you wish to further increase the default capacity of your numbers.
The 2 default channels also serve as a guarantee that even if all flexible channels allocated to the group (pool) are occupied, each of your DIDs will still have the capacity to receive (at least) two concurrent incoming calls. Dedicated (non-shared) channels are allocated to a specific DID, and allow additional simultaneous calls on this particular single number. By default each number provided by Handility is equipped with 2 voice channels; therefore, additional channels are necessary only if you wish to further increase the default capacity of your number. Dedicated (non-shared) channels are assigned to a specific DID, and allow additional simultaneous calls only on that particular number. Note, that by default, all DIDs are provided with two dedicated (non-shared) channels at no extra cost. But you may increase the concurrent call capacity of your selected numbers by adding extra channels. Flexible (shared) channels cover a group of selected DIDs, and allow additional call capacity on all those numbers. As additional incoming calls are received, flexible channels are dynamically allocated to DIDs within a specified group on a first-come, first-served basis. The group of numbers, flexible (shared) channels are assigned to, may be comprised of DIDs from anywhere in the world, for example, DIDs from Canada, Spain and Australia may be included in the same flexible channel group. It should be noted that, by default, all DIDs are provided with two dedicated channels, over and above any flexible channels purchased. This ensures that even if all flexible channels allocated to a group are occupied, your DIDs have the capacity to receive (at least) two concurrent incoming calls. |
| FAQs about handility cloud phone system, drag and drop phone system functionality and use Unfortunately due to the nature of our system, in order to ensure security and reliability we currently DO NOT support incoming calls from other providers to the Cloud Phone System. However this does not limit you from using the full functionality of our system. You can either:
Sure you can. Contact us for more details. Unfortunately Skype has gone through a policy change and no longer supports open connectivity with outside services such as ours. Skype has become a proprietary network locked to its own software. For more information, please visit http://en.wikipedia.org/wiki/Skype_protocol This is one of the most revolutionary features of Handility.We all know that when traveling abroad mobile phone roaming charges can be very high. Receiving and placing calls doesn't feel as easy as when at home. With Handility phone number as part of our service you can eliminate roaming charges in over 60 countries Step-by-step Instructions
When DIDs are mapped to the Hunt Group, all destinations configured under this group will ring in order one after another with a timeout configured per destination, you can add 2 or 3 mappings. Once destination is answered, the call is connected and call hunting is stopped. This group is useful if you wish to be reached everywhere you go, or if you would like to have priority for the people who answers the phone. For example: The call first will try to reach you at home, after 15 seconds it will try to reach you at work and after 15 seconds - on your cell phone. To create Hunt Group please follow the below steps: Hunt group is an option allowing you to map your DIDs to up to 3 different destinations that will all ring in order, one after another, following your specified time outs. This is an indispensable feature for your call redundancy and backup. Using a hunt group you can be reached in up to 3 different locations or on up to 3 different devices, e.g. the call will first reach your home land line phone, after 15 seconds it will be redirected to your office and after another 15 seconds will ring on your cell phone. This option also comes in handy when you need to set preferences and priorities for who is answering the phone in your company first. Once any of the available destinations is answered, and the call is connected, all other destinations cease ringing. |
| About connectivity, types of devices connected, number mapping, forwarding to a PSTN in a different country and more... Configuration is done via a simple, user-friendly menu on our website, where the customer is provided with a selection of forwarding options to choose from. Yes! In addition to offering Direct Inward Dialing numbers (DIDs) which are non-portable standard 10-digit (including area code) numbers, we also offer portable numbers, and can arrange to port-in your number from most major telephone providers. We can also help you port the number out, should you decide not to use Handility services any longer. While the speed of the Internet connection isn't critical, the key factors that affect call quality are reliability and some available bandwidth. Make sure you choose an ISP with a guaranteed up-time to ensure a consistent and reliable connection. As our system is connected to a super fast, highly reliable backbone network, however every system is only as weak as its weakest link, so a slower, unreliable connection on the client side can sometime cause a bottleneck. To manage your bandwidth we recommend you have an ADSL2+ modem that supports Quality of Service (QoS), which will make sure that your computer's downloads don't affect your phones. Alternatively cable internet or satellite internet with high-availability is a good option.
Yes. In case you wish to change your PSTN number, please note that country should be supported by the same Zone. In case the country is included to different Zone, new PSTN plan should be ordered. IN addition, if a number previously forwarded to a non-PSTN device (such as VOIP etc) now should be forwarded to PSTN (landline/mobile phone number), then an additional flat monthly charge is applicable. DTMF (Dual Tone Multi Frequency) is the signal to the phone company that you generate when you press an ordinary telephone's touch keys. The call records in the Calls page are displayed in GMT time zone. The DID number will be forwarded to the mapping you created immediately, please make sure that your mapping is correct.To see how to map existing DID numbers please see How to Map Existing DID Number?
You can see the list of Handility IP addresses at The List of Handility IP Addresses. Mapping is the connection between your DID number provided by Handility and your call forwarding destination, which can be any of your chosen mobile or land line numbers, VoIP providers etc. You can use the following mapping options to forward your calls:
For information on how to create the mapping see How to Add New Mapping? Mapping is the connection between your DID number provided by Handility and your call forwarding destination, which can be any of your chosen mobile or land line numbers, VoIP providers etc. The easiest way to go about creating a new mapping and configuring your number is as follows: select the type of mapping that suits you most and is compatible with your system (for further information please refer to Which Mapping Option Suits Me Most?); To view the list of mapping options available, please enter your account and proceed to the section Mapping Yes.Most countries allow you to forward calls to both regular PSTN (land-lines) and mobile phones. Yes. We support call forwarding to land line and mobile phone numbers. Yes you can purchase DID number with PSTN mapping and forward the calls to any country. In case the country is not listed, you are welcome to use Pay As You Go plan which allows to receive calls anywhere in the World. This function allows you to receive calls to your DID numbers on regular land line or mobile phones.No internet, PC or VoIP services are required and no minutes are charged. You can always login to your account and change the mapping.Just please note that if it is PSTN mapping, the country should be of the same Zone as your current one. In case this country is included in another Zone, another PSTN plan should be ordered. Remember, the number must be entered in country-code+city-code+number format. The countries to which call forwarding to PSTN is supported, are listed in the Mapping page under PSTN. Yes, as long as your VoIP provider supports our service.Please make sure that your VoIP provider is in Handility predefined list, if you cannot find your VoIP provider in the predefined list please send our support team the information requested. |
| Questions about VOIP, SIP phones, softphones, codecs and mapping calls to VOIP as well as pre-configured providers Handility can serve as a VOIP service or as a layer "on top of" existing VOIP.Handility allows DIDs to be mapped to a number of 3rd party VoIP providers. The selection of the VoIP provider is performed by using our on-line configuration interface. The following is an example for setting 3rd party SIP services at Handility.If you have an account at SIP service provider such as CallCentric (http://www.callcentric.com/) and your account number is: This email address is being protected from spambots. You need JavaScript enabled to view it. your mapping should look as the instruction for How to Set VoIP Mapping for CallCentric below. If your VoIP provider is not listed in the VoIP provider list at the VoIP mapping page, please open a support ticket request with your provider’s details as specified in Add your VoIP Provider to the Predefined List or set Custom / URI mapping as instructed in Can I Map VoIP Provider that is not In the Tested Providers in another Way? To ensure the CODEC for calls from Handility you need to set at your system the desired CODEC as the only CODEC for this connection. For example, if you wish to use ULAW CODEC only (on Asterisk) you need to configure a trunk to Handility and make sure that the following lines are added to the trunk:
You should open port range 10,000 through 20,000, this is the default RTP ports range used by Asterisk. Yes.Handility supports DNS SRV (Domain Name Server Service) The IP addresses you will receive calls from depend on the preferred server you chose in the mapping you created for the DID number.If you chose US as preferred server, the call will be sent from USA range of addresses:
In addition, RTP media may arrive from 46.19.209.0/25, please allow the range in your firewall.
For Asia and Australia RTP media may arrive from 46.19.210.0/25, please allow the range in your firewall. Yes, it is possible.For this purpose you can use the preferred server option. If you set the US server, for example, as preferred in your mapping settings all calls will be sent to you from the US server. Please try using Inband or RFC2833 DTMF type on your endpoint to resolve the DTMF recognition issue. You need to create an incoming dial peer as follows:
Where 12345678 is your DID. Please note that if you have an access-list on the Cisco gateway, do not forget to permit the traffic coming from the Handility IP addresses. To see the list of Handility IP addresses go to The List of Handility IP Addresses.
Please find more information about Handility Configuration Scripts Handility supports the following CODEC types:
Yes. It is possible to map VoIP provider that is not in the tested VoIP providers in other ways. You can use those ways, for example, if Handility tests were failed with this provider but you still want to use it.
There are two ways to add VoIP provider that is not in the list:
Handility supports RFC2833.
No. Handility supports both dynamic renewable (using hostname) IP and Static IP for VoIP phones. The following list contains the VoIP providers that failed Handility phone.systems tests and cannot work with Handility virtual numbers:
If your VoIP provider is not listed in the VoIP provider list please open support ticket with your provider’s details as specified in Add your VoIP Provider to the Predefined List or set Custom / URI mapping as instructed in Can I Map VoIP Provider that is not In the Tested Providers in another Way?
The following is the most updated list of VoIP providers which support Handility services:
If your VoIP provider is not listed in the VoIP provider list please open support ticket with your provider’s details as specified in Add your VoIP Provider to the Predefined List or set Custom / URI mapping as instructed in Can I Map VoIP Provider that is not In the Tested Providers in another Way? Handility holds a list of all tested and approved VoIP providers.Please note that occasionally service providers change their settings, the list will be updated accordingly. In order to add your VoIP Provider to the predefined list, please open a support ticket with the following information:
a. The web Site of the VoIP Provider. b. IP Address or domain of the VoIP Provider. c. VoIP Protocol type used. d. Account username, registered at the VoIP Provider Please note that without account username, IP Address or domain and Protocol type, Handility cannot check the service and the inquiry will be canceled. Please also note that not all VoIP Providers accept calls from external networks. Therefore, not all VoIP providers can be added to predefined list.
Yes, we support IAX. The following information will help you to set the IAX with the Handility IP addresses:
Create context on IAX.conf:
[46.19.209.10] type=friend host=46.19.209.10 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.11] type=friend host=46.19.209.11 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.12] type=friend host=46.19.209.12 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.13] type=friend host=46.19.209.13 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833 [46.19.209.14] type=friend host=46.19.209.14 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.15] type=friend host=46.19.209.15 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.42] type=friend host=46.19.209.42 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.75] type=friend host=46.19.209.75 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.76] type=friend host=46.19.209.76 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.77] type=friend host=46.19.209.77 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.78] type=friend host=46.19.209.78 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.79] type=friend host=46.19.209.79 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
[46.19.209.80] type=friend host=46.19.209.80 trunk=yes context=from-Handility qualify=no canreinvite=no dtmf=rfc2833
Under 'iax_general_custom.conf.' add following lines: requirecalltoken=no calltokenoptional = 46.19.209.0/255.255.255.0 Network Address Translation (NAT) is a common practice used in networks, and it doesn't play well with VoIP. Solving this problem requires an understanding of NAT, VoIP and your VoIP setup. The following focuses on the SIP protocol for VoIP using Asterisk, but problems and solutions are applicable to most other situations. NAT can cause problems in several places. If one of the PBXes is behind a NAT gateway, the other PBX will not be able to contact it without some additional network setup. If one or more of the phones are behind a NAT gateway, the other phone will be trying to send audio to a non-routable address. This results in failed calls or missing audio.
The alternative to a re-invite is to have the PBX relay the voice packets between the two endpoints. SIP client is behind a NAT gateway connecting to a server on the Internet
The client creates the translation entry for the SIP traffic when it first registers. As long as there is frequent communication between the two hosts, such as one packet per minute, the channel will stay open. The only configuration needed is to have the client use its external address in all SDP packets. On clients that support it, enable STUN (Simple Traversal of UDP through NAT), so the client can determine the external address dynamically, or enter it manually. Asterisk doesn't support STUN at this time, so all NAT configuration must be done manually. The following commands in /etc/asterisk/sip.conf set up the NAT properly:
[general] localnet=192.168.0.0/255.255.0.0 // or your subnet externip=x.x.x.x // use your address [YOURREMOTEPEER] // your peer's name nat=yes qualify=yes // force keepalives
With this configuration, Asterisk uses the address defined by externip for all calls to the peers configured with nat=yes. The addition of qualify=yes causes Asterisk to test the connection frequently so that the NAT translations are not removed from the firewall. With these two commands, there always will be a communications channel between Asterisk and the peer, and Asterisk will use the outside address when sending SDP messages.
Multiple SIP phones and an Asterisk server behind a NAT gateway
Calls between the phones will work fine because NAT is not needed. For calls between you and other systems on the Internet there will be problems. Unless you register to the remote side as a client (as done in the previous example), you will not be able to receive SIP messages, so you will not be able to accept calls. Second, the address information in the call setup will point to the internal address of the phone, and the one-way audio problems mentioned previously will crop up. The easiest solution to this is to avoid NAT entirely. If you have a public IP address available for your call server, use it. If your Asterisk server is connected to both the Internet and the internal network, the SIP port is reachable from both the inside and the outside, and the only problem is ensuring RTP flows properly. The PBX server does not need to be configured to route between the interfaces or provide masquerading; it simply needs to bridge the inbound and outbound voice calls. As I mentioned earlier, the PBX either can stay in the voice path or get out of the way. In the latter case, the PBX tells both endpoints about each other after which the endpoints talk directly. However, Asterisk could have a call setup with both endpoints and relay the RTP packets on behalf of each endpoint. The inside host would be talking to the inside address, and the outside host would be talking to the outside address. The only configuration required to achieve this in sip.conf is to disable re-invites: [general] canreinvite=no // force relaying This configuration works well because the Asterisk server can speak freely to the Internet to send and receive calls. It also can talk to the internal phones, and by some simple bridging, completely ignore NAT. As it turns out, this relaying behavior also is required when the Asterisk server has only a private address. The RTP ports will have to be forwarded on the firewall too. RTP chooses random port numbers based on configured limits. Before the ports can be configured, they should be limited in range. Configuring the firewall rules is much easier if the range of ports is known beforehand. The range of ports to be used for RTP is defined in rtp.conf. The following configuration will limit Asterisk's choice of RTP ports from 10000 to 20000:
[general] rtpstart=10000 // first port to use rtpend=20000 // last port to use, rounded up if odd Asterisk will need several RTP ports to operate properly. Only even ports are actually used, and disabling of re-invites causes two connections to be built per call. These ports and the SIP port must then be forwarded in by the firewall. The iptables syntax is: iptables -t nat -A PREROUTING -i eth0 -p udp \ -m udp --dport 10000:10100 -j DNAT \ --to-destination 192.168.1.10 iptables -t nat -A PREROUTING -i eth0 -p udp \ -m udp --dport 5060 -j DNAT \ --to-destination 192.168.1.10
Replace eth0 with the outside interface of your firewall and 192.168.1.10 with the address of your Asterisk server. These rules tell the Linux kernel to translate the destination address of any UDP packets in the given range that are entering the outside interface. This must happen at the PREROUTING stage as opposed to the POSTROUTING stage, because the destination address is being translated. At this point, any SIP or RTP packet from the Internet will be forwarded to the internal Asterisk server for processing. When a remote station makes a call to Asterisk, the SIP packet will be forwarded in because of the iptables rules. Asterisk will stay in the media stream because of the canreinvite=no command and it will use the external address of the firewall in any SDP packets because of the NAT commands. Finally, the media stream will be forwarded to the Asterisk server because of the combination of iptables RTP forwarding and port ranges defined in rtp.conf. Up to this point, the configuration has focused on getting Asterisk working behind a NAT gateway, with some extra details to make the phones relay through Asterisk. There are, of course, more general solutions. If you can avoid NAT in the first place, it is in your best interests to do so because it avoids all the problems encountered so far. The Asterisk gateway can have a very restrictive firewall policy applied to it – you just need to allow UDP 5060 for SIP and whatever port range is defined in rtp.conf. In this configuration, Asterisk can contact both the internal phones and the rest of the Internet. If SIP is not a requirement, and you are using Asterisk, consider using the IAX protocol. IAX tunnels both the control traffic and the voice traffic over a single UDP conversation that can be port-forwarded, filtered or translated easily. This method is limited to a static set of tunnels, which is sufficient if you are connecting some PBXes over the Internet or connecting to a long-distance provider. Sometimes the above solutions are not available to you. In that case, it might be advisable to move to a full-featured SIP proxy and use Asterisk only for voice applications, such as voice mail. SIP Express Router (SER) is a powerful SIP server that handles NAT well and is used by several high-volume services, including Free World Dialup. SER's job is only in setting up calls between endpoints, so it must rely on other applications, such as specialized media proxies, to handle RTP streams if needed. The step beyond a SIP proxy is a Session Border Controller (SBC), which is like a VoIP firewall. The SBC can intercede in either the signaling or RTP paths to add extra features, such as signaling protocol or codec translation, all while enforcing security policies. These are almost exclusively commercial products. Please view the explanation in How to Set SIP Configuration for Handility DID Number on Asterisk? for defining the trunks in an Asterisk PBX system. Please be aware that "allow all" refers to all kind of CODEC (e.g. ILBC or G729 CODEC). For the example purpose you have 3 numbers on your Handility account, as follows:
And you mapped them to your SIP server with the address 1.2.3.4. The numbers will be forwarded to your SIP server as follows: Your server will identify each call by the called number (extension). Yes.Handility.com allows DIDs to be mapped to a number of 3rd party VoIP providers. The selection of the VoIP provider is performed by using our on-line configuration interface. Yes.It is possible to forward an incoming DID number to the SIP Server of your VoIP provider as long as this provider is on the predefined list. If your VoIP provider is not in the predefined list please follow the instructions as specified in Add your VoIP Provider to the Predefined List. Give Trixbox or Asterisk a try. There is a slight learning curve, but both are well documented. These provide an alternative to 3CX (windows based) system. 3CX is well documented as well. Yes.Handility system accepts non-standard SIP ports. The following guide will explain how to set new DID number on Asterisk.The following guide will explain how to set new DID number on Asterisk. In the following explanation we will use DID number 972775654199 for example, this DID number will be forwarded to the internal SIP extension 100. Your Asterisk configuration files should look as follows:
sip.conf: [46.19.209.10] host=46.19.209.10 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.11] host=46.19.209.11 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.12] host=46.19.209.12 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.13] host=46.19.209.13 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.14] host=46.19.209.14 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.15] host=46.19.209.15 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.75] host=46.19.209.75 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.76] host=46.19.209.76 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.77] host=46.19.209.77 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.78] host=46.19.209.78 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.79] host=46.19.209.79 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.209.80] host=46.19.209.80 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.10] host=46.19.210.10 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.11] host=46.19.210.11 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.12] host=46.19.210.12 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.13] host=46.19.210.13 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.14] host=46.19.210.14 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.15] host=46.19.210.15 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.75] host=46.19.210.75 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.76] host=46.19.210.76 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.77] host=46.19.210.77 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.78] host=46.19.210.78 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.79] host=46.19.210.79 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
[46.19.210.80] host=46.19.210.80 dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-Handility insecure=very nat=never allow=all
extensions.conf: [from-Handility] exten=> 972775654199,1,Dial(SIP/100) Please view the explanation in How to Set SIP Configuration for Handility DID Number on Asterisk? for defining the trunks in an Asterisk PBX system. Please be aware that "allow all" refers to all kind of CODEC (e.g. ILBC or G729 CODEC).
You can generate the ring tone by adding/changing the following strings:
exten => s,1,Answer exten => s,n,Dial(SIP/andrej,,r) You need to add entry to the SIP.CONF of the Asterisk server each time we add new IP addresses to our servers. You can see example in our knowledgebase: How to Set [email protected]/TrixBox with Handility DID Numbers? SIP.CONF needs to be modified every time we add IP address (you will need to change only the name of the entry section and the “host=” parameter). Our servers send DTMF packets with a different duration than is set in NMS Communication software (e.g.: 160 instead of 0 and 1440 instead of 800).
At the NMS you should set that all durations will be accepted. If you have upgraded Asterisk to 1.4 version you might have problems with the DTMF. To fix this issue you should add to sip.conf the parameter 'rfc2833compensate=yes' to fix the DTMF issue. Problem: When the application (Asterisk based) asks user to enter a phone number, some of the digits appear twice, it happens in about 70% of cases. For example, if user entered a number: 9542322800 then the application will receive: 955423228000 or another combination where one digit or few is captured twice. Answer: Please try to use Inbound or RFC2833 DTMF type at your endpoint to resolve the DTMF recognition issue. Furthermore, you can try another carrier to see if this issue is related to your carrier. If the problem was not solved please open a support ticket. Problem: When dialing the DID number I can connect to the server but when I enter the PIN code or my phone, some digits are not received in my server, as a result I cannot make calls. Answer: Please check which VoIP protocol you are using, if you are using H.323 protocol you should make sure that you configured rfc2833 as your default DTMF mode, if it does not work please try to change it to h245signal. If the problem is not solved please open a support ticket with PIN code for testing. |
| A collection of step-by-step guides and video tutorials No faqs found in this category
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| FAQs related to payments, account info, billing and more... This refers to Local Access Number (your local or tollfree number purchased from Handility) Subscriptions can be cancelled anytime in the customer area. There are no minimum contracts. You can pay for your number or plan for 30 days and cancel any time. You may purchase an extended plan for 60-90 or an annual plan in order to receive significant discounts. We designed Handility to work on subscription basis with a pre-paid balance model. As a Handitliy user, we recommend you keep a positive balance on prepaid account to acquire and use our services (such as phone numbers, per minute charges or other added services you may wish to add and use). You do have total control over your cost (by viewing your account page on handility.com and only pay for what you use) You can see the monthly statement for your services on your monthly invoice. A monthly invoice can be found in the Billing section of your Handility User Web Control Panel. Go to handility.com >> Account No. You will not be able to receive calls; therefore, please make sure your prepaid balance is positive. If this particular feature is a problem for you, your business or operation, please contact us at This email address is being protected from spambots. You need JavaScript enabled to view it. and we may be able to set-up a solution that suits your needs The setup fee is paid only once when you purchase the first DID number. We apply the setup fee only once per customer - the fee covers administrative and technical cost of making sure the system is stable, the number is connected and the payment information is set up correctly.
We offer a 30 day money back warranty. This usually applies to the monthly subscription fee. The only portion of the service which is non-refundable is the set-up fee as well as any additional per minute cost you have used. We offer a 30 day money back warranty. This usually applies to the monthly subscription fee. The only portion of the service which is non-refundable is the set-up fee as well as any additional per minute cost you have used. The payment is processed via credit card (Visa, MasterCard, American Express and Diners Club International). We also accept Paypal. Additional payment methods are available, please contact your account manager or our customer service by e-mail This email address is being protected from spambots. You need JavaScript enabled to view it. . Handility provides users with numbers at fixed monthly rates, starting at under $5/month. Setup fee is applied for first month only. Monthly rates depend on the country in which the local telephone number is purchased (local, tollfree, etc) Each regular DID is billed on a fixed monthly fee basis, with no per-minute charges. This is a flat-rate service, which means that you can talk as much as you want for a low, monthly charge.
Handility.com accepts US Dollars only at this time for all transactions. For prices and transaction billing in Australian dollars, please visit handility.com.au All international Credit Cards should allow you to purchase services in US Dollars and have it converted to your local currency on your monthly statement (NOTE: Your Credit Card issuer sometimes charges small fees for this conversion). |
| We'd love to help you anytime. Here are a few easy troubleshooting tips to help you get started... Make sure you are dialing correctly. All calls must be dialed in full E.164 format whcih is full country code and number. There are no prefixes needed that you may be used to. A few samples of how calls should begin: US: 1 + area code + number UK: 44 + number Australia 61 + number You may setup your softphone (Zoiper, X-lite) or the Phone.Systems app to automatically select your country code - in order to expedite the process of dialling locally. First, please check the following:
If you receive calls on a regular PSTN phone - you set number forwarding to PSTN
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| Statutory requirements, legal restrictions (country specific), limitations, fair use, privacy, data use When you purchase a DID via our online 24/7 web services, activation is instantaneous. Nevertheless in some cases of unverified details the number will be activated only after verification of Handility Customer Care team. In such case the our Customer Care team will contact you in 24 hours. However, service is working, while it is in pending status. Some local carriers require end user's information for registration before purchasing a DID(local or national phone number)While purchasing the number you will be asked to fill contact information in a form. After completing the purchase your information will be transferred to the local carrier for review. Please note that your DID might not be activated until the local carrier approves the details.*
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Great product. Great service. I havent had issues with support. Its plug and play and simple to setup. No landline phones for two years now. VOIP is the only way to go!
John Karterados, Boutique Clothing Store Owner, Member since Jan 2015
Being able to reuse our current office phones was a big one for us, and Handility offered us one of the best solutions in this regard. Hooking different people up to the system so they can set up their smartphone and office phone as they want works great and made all of our lives easier. The online faxing is a welcome extra to us since we never had a fax before but sometimes suppliers request us to send or our orders through fax.
Jane Chi, Consultant working with clients Worldwide, Member since mid 2014
Very glad I committed to the Handility solution. I’ve been without a traditional phone line for three years (owning just an Android phone), but looked back at a few options to get a local number again. Now I’m using the Handility mobile app, got a Voip phone as well for home and just started using their App. I used another service before but they were lacking in technology and they were more expensive. These guys are top notch. I’ve recommended them to my brother (who’s self employed) and to my best friend, who started a HVAC company.
Jerry, Warehouse Manager, Member since Oct 2014
It is a breath of fresh air to be able to set up a professional recording, voice forwarding, interactive menus, fax to email, voicemail to email etc - all for under 10 bucks per month. For a start-up like ours this just makes sense.
Monica, Online Store Start-up owner, Member since March 2015
