Теле2 sip trunk
SIP Trunking - voip-info.orgFrom the SIP RFC 4904:
A Session Initiation Protocol (SIP) to PSTN gateway may have trunks that are connected to different carriers. It is entirely reasonable for a SIP proxy to choose — based on factors not enumerated in this document — which carrier a call is sent to when it proxies a session setup request to the gateway. Since multiple carriers can transport a call to a particular phone number, the phone number itself is not sufficient to identify the carrier at the gateway. An additional piece of information in the form of a trunk group can be used to further pare down the choices at the gateway. As used in this document, trunks are necessarily tied to gateways, and a proxy that uses trunk groups during routing of the request to a particular gateway knows and controls which gateway the call will be routed to, and knows what trunking resources are present on that gateway.
In an architecture where calls can be terminated on multiple gateways it is wise to consider routing the call to a destination based on some significant criteria such as cost, quality or proximity. Where a proxy has the ability to evaluate a call based on one or more of these criteria, as well as knowledge of the TDM trunk resources available, the proxy can "tag" the call using the tgrp and trunk-context values in the SIP Contact field of the INVITE. It is important to note that the tgrp and trunk-context values can only be used with a TEL URI, not with a SIP URI.
Unlike in traditional telephony, where bundles of physical wires were once delivered from the service provider to a business, a SIP trunk allows a company to replace these traditional fixed Public Switched Telephony Network (PSTN) lines with PSTN connectivity via a SIP trunking service.
What is SIP Trunking and how will it help my business?Basically, SIP Trunking is a service that provides VOIP or Voice over Internet Protocol. In other words, it is a form of communicating by transmitting telephone calls over the Internet. This communication through the Internet is done by connecting the private branch exchange (PBX) to the Internet. The Internet actually replaces the telephone trunk allowing for communication by users with both fixed and mobile telephone subscribers throughout the world.Your voice, data and videos are all combined into a single line with Session Initiation Protocol (SIP) Trunking. This allows for your local, long distance and broadband Internet service to be combined into one line. You will be able to keep your real time traffic off the internet as well as off the public switched telephone network (PSTN) as much as conceivable.
Advantages of SIP TrunkingThe advantages of SIP Trunking over traditional telephone lines and older VOIP protocols are several:
- Whereas before SIP you needed to carry voice, video and data over one line by using a Primary Rate Interface (PRI), the SIP Trunking eradicates the necessity for gateways of Basic Rate Interfaces (BRI), Primary Rate Interfaces (PRIs) and PSTN.
- The provisions of incoming, outgoing and Private Branch Exchange (PBX) are made by your VOIP business provider setting up a proxy server also known as a SIP proxy.
- Your provider also does all technical support. This saves you both time and money since you will no longer need an IT team or an IT contractor.
SIP Trunking Saves You MoneySIP Trunking allots lower costs without sacrificing quality. When it comes to pricing, SIP Trunks are significantly cheaper than the customary analog circuits. What is the saving? The cost of SIP trunks will range from approximately $20 to $30 per trunk, whereas the analog circuits cost roughly $30 for each circuit. There are also significant savings with charges of long distance terminations with SIP Trunks costing considerably less than TDM rates or customary analog rates. All calls are local with SIP. The result for your customers is that both incoming and outgoing calls have an area code that is local. This gives you a lower cost for your business, and your customers get a feeling of familiarity and closeness with your business. The cost of SIP calls per minute are only a fraction of a penny. In addition, SIP numbers that are toll-free are also available to you.
Another factor that can be costly for your business is if you want to up the number of Primary Rate Interfaces (PRI) from 23 to 24 channels, you must buy a second PRI that has 24 channels. This would, therefore, give you 20 unused channels. With SIP, it is possible for you to add, subtract or move a line by simply making arrangements with your provider. This saves you cost by never requiring you to have more or less lines than you actually need. Even if you should have to move your office, with SIP, you are able to keep all the same internal extensions and inbound numbers. Basically, SIP Trunks cater to your needs by allowing you to purchase in additions of one instead of overloading your office with costly, unnecessary additions.
Easy Implementation and Integration with Existing SystemWith the use of an Integrated Access Device (IAD), the SIP-ready PBX required can be easily modified to function with key or customary analog systems. The necessity of PRIs, BRIs, or local PSTN is completely eliminated with SIP as I mentioned above.
As systems become more complex, it is recommended that you increase the speed and capacity of your Internet. If you move E1 and T1 Interfaces to two, you will need additional hardware.
Multi-site businesses may not communicate via multiple sub PRI connections. Instead of the multiple sub PRI connections, they may communicate via a single SIP Trunking account.To build an infrastructure will require you to purchase an IP-PBX, IP enabled devices, IP Phones, or softphones (software telephones that make it possible to place voice calls through your computer and over the Internet) all of which are relatively inexpensive.
Another wise step you should take is to put a firewall that is SIP-aware in place in order to regulate your VOIP traffic. The return on investment (ROI) of these systems is normally both obvious and fast, and the switch will be well worth the minimal cost and time. You certainly won’t regret making the switch.
Emergencies that may cause the disabling of your communications at your location are easily overcome:
- Via predetermined failover services, your SIP trunks will automatically redirect calls to additional locations and extensions. For instance, should a power outage occur, your calls will be automatically redirected to a satellite office.
- It is also possible for many businesses to set forwarding options to their smartphones since the use of mobile devices and the practice of bring your own device (BYOD) have grown so rapidly in today’s business world. Mobile devices can be a good backup if your primary communications go down. In addition, since a number of these systems are optimized for networks that require low bandwidth use of data or Wi-Fi, the calls most likely will be totally free.
SIP Offers FlexibilityIf you’re looking for outsourcing or your company has an international presence, SIP can meet your needs. For example, SIP can spread your customer service staff across the country or even around the world. Therefore, SIP makes it possible for your business to be open 24 hours a day and 7 days a week.
There is no need for a phone line at all with SIP. A Session Initiation Protocol Uniform Resource Identifier (SIP URI) address includes numbers and the name of the host. This system entirely bypasses the PSTN and has no cost at all to connect to a different SIP number. Many workers whether on-site or remote prefer to use their own IP-Enabled devices for SIP. These IP-Enabled devices include PDAs, tablets, smartphones and laptops.
What you don’t know about ?SIP trunking is that it combines data, video and voice networks into one line, including your broadband, long distance and local calling as well. This means that you can keep all of your real time traffic off of the PSTN and on the internet instead, which has plenty of advantages over the traditional telephones or even the older versions of VOIP.
Highlights of SIPSIP offers small businesses the opportunity to appear both large and professional, and large businesses benefit by maintaining a local presence. Whether your business is small or large, the most significant feature of any system is for it to be reliable and able to add to your bottom line. VOIP business services and specifically SIP will do that for you.
The pricing, services and features of SIP Trunking vary greatly with the different providers.
To be added...
- PSTN-to-PSTN calls traversing SIP networks
What is SIP Trunking?
SIP Trunking helps businesses save money by using their existing intranet or internal communication system to access the internet and outside phone lines. This system replaces a traditional phone system. SIP Trunking directs incoming and outgoing calls or data to an internet phone provider or the company’s private network.
The Basics of SIP
SIP is an acronym for session initiation protocol. Session initiation protocol gives users control over information sent from a private branch exchange (PBX) through an internet protocol (IP). The IP can be an internet address or voice over (VoIP). It also takes information from an internet telephony system provider (ITSP) and tells the PBX where to direct the information.
When communicating with a business’ PBX, the SIP:
- Keeps up with the location of each line.
- Tells the ITSP and the PBX the expected action (ring this phone line, send this busy signal)
- Determines if a line is free, in-use or unavailable
- Directs information where it is supposed to go (i.e. voicemail, call waiting, fax etc.)
- Tells a phone system which number was dialed and from where.
- Allows both systems to know when a call has ended or been transferred.
For external calls, usually a public switched telephone network (PSTN) handles these parameters for businesses and residences. Having a system that keeps up these parameters lets a business do more with their existing system without paying more.
Using a SIP Trunk Increases Productivity
SIP trunking increases productivity by adding more functions to an existing system. With SIP trunking, businesses can:
- Make calls from their intranet, including local and long distance.
- Send emails, text messages and videos externally.
- Conduct internet searches on their existing system
- Have access to directory assistance and emergency 9-1-1.
Without an SIP trunk, a company’s intranet is only able to communicate inside the company unless it uses a PSTN’s hardware (PRIs). With a SIP trunk, a company uses the VoIP to connect to outside networks using their phone system.
SIP Trunking Doesn’t Require Extensive Equipment
This service requires a VoIP phone system that understands the information provided by the trunk. Older analog systems can still connect using a SIP gateway, but the business would not get the full benefit of the SIP. SIP gateways translate information into contexts an analog phone system understands. Since the protocol provides real time data information, such as video images, caller ID or call waiting ID, businesses miss out on these benefits using analog systems.
Businesses need a router capable of prioritizing information presented by the SIP. The router not only connects the system to the internet, but it also determines what goes first — phone calls or emails. A router that sends information on a first come first serve basis does not take into account that text messages or other data are not as important in real time as conference calls.
Companies that have a PBX with a SIP trunk can use their system to connect to an ITSP. If this investment has not been made, the ITSP can host a full-featured PBX. In some cases, companies with a PBX but few users can affordably switch to a hosted PBX and avoid maintenance, monitoring and manpower required to keep up the system.
SIP Trunking Requires Three Services
As mentioned, the first service needed is anITSP. The ITSP acts as the switchboard operator. It receives and sends phone calls and data between the business’ IP address and an external IP or phone line. The ITSP also provides telephone numbers for the business.
The second service is aqualified installer. The business is responsible for setting up their internal systems. This includes getting the PBX, router and phone systems to send and receive information internally as well as to and from the ITSP.
The third and last service isinternet. SIP trunking eliminates the use of traditional public phone company wiring. Without a PSTN, companies rely heavily on their internet service for their phone and data needs.
A SIP Trunk Must Have Quality Internet Access
Without including the bandwidth savings of compression software, businesses need about 80kbps per line for optimal service conditions. Broadband services such as fiber optic or cable lines offer quality access, but pay particular attention to upload and download speeds. Often services have high speed reception, but the upload speed is not comparable.
The SIP trunk sends and receives data quickly. Slower upload speeds delays the messages and sounds the person at the other end of the line receives. In order to be sure enough bandwidth is available, companies can examine existing line usage for comparison. Since compression software frees about 30% of bandwidth usage, buying the optimal bandwidth amount allows room for growth, i.e. additional lines and services.
SIP Services Decrease Business Phone Expenses
By using SIP to access the ITSP, businesses save money. Since every phone must have a line for PSTNs, businesses eliminate the cost of upkeep for these lines. If the business has more than one primary rate interface (PRI) bundle, the SIP completely eliminates this expense. As long as the bandwidth is available, additional lines are easier to add at much more affordable rates.
Businesses save on long distance costs as most phone calls within the country are inexpensive or free, and international phone calls cost less than most PSTN fees. The SIP trunk connected with an ITSP provides more features than PSTNs without additional costs. Here are a few of the services this system provides that PSTNs traditionally charge for on a per-line basis:
- Caller ID
- Call Waiting
- Dedicated Fax Line
This protocol improves scalability by taking away the 23 voice channel increments and eliminating PRI maintenance costs. Virtual telephony services offer built-in resources for redirecting calls during internet outages or other disasters, which makes disaster recovery plans implementable. SIP trunking is transferring the way a business accesses the outside world from physical to virtual connectivity.
IP Çoklu Hat Hizmeti | Türk Telekom Kurumsal
IP Çoklu Hat Esnek Tarife ücretlendirme bilgisi
|0 - 9.999||0,068 TL||0,085 TL|
|10.000 - 19.999||0,060 TLL||0,075 TL|
|20.000 - 29.999||0,056 TL||0,070 TL|
|30.000 - 39.999||0,053 TL||0,067 TL|
|40.000 - 49.999||0,052 TL||0,065 TL|
|50.000 - 74.999||0,051 TL||0,064 TL|
|75.000 - 99.999||0,050 TL||0,062 TL|
|100.000||0,049 TL||0,061 TL|
IP Çoklu Hat DID (İç Hat) Ücreti
|0,83 TL||0,38 TL||1,04 TL||0,47 TL|
IP Çoklu Hat Kanal (Dış Hat) Başı Sabit Ücret
|Kanal Başına||11,05 TL||13,87 TL|
Yukarıda belirtilen ücretler aylık olarak faturaya yansıtılan ücretlerdir. Dakika ihtiyacınıza göre dakika birim ücreti yukarıda belirtilen denk geldiği barem aralığındaki dakika birim ücretine göre ücretlendirilmektedir. Hacim bazlı tarifeyle artık paket aşımı takibi de yapmak zorunda değilsiniz! Aldığınız dakikayı aştığınız durumda dakikanızın denk gelen bareminin dakika birim ücretinden faturalandırılmaya devam edersiniz!
IP Çoklu Hat Kanal (Dış Hat) Başı Bağlantı Ücreti
|Kanal Başına||15,94 TL||20,00 TL|
Yukarıda belirtilen bağlantı ücreti ilk ay olmak üzere tek seferlik olarak faturaya yansıtılır.
Örnek hesaplama aşağıdaki gibidir:
75 Kanal, 155 DID, 60 Rezerv DID, 31.300 toplam dakika ihtiyacınız varsa ücretlendirme aşağıdaki gibidir:
(75*14,70 TL)+(155*1,1 TL)+(60*0,50 TL)+(31.300*0,071 TL) = 3.509,80 TL
İş Avantaj Her Yöne/Kamuya Özel Her Yöne Tarifeleri
IP Çoklu Hat Ürünü kapsamında barem yapılı esnek tarifeye ek olarak mevcuttaki İş Avantaj Her Yöne ve Kamuya Özel Her Yöne tarifelerinden birini seçerek de Türk Telekom’un yeni nesil ses hizmetinden faydalanabilirsiniz! Esnek yapısı sayesinde her bir kanala farklı bir İş Avantaj Her Yöne veya Kamuya Özel Her Yöne tarifelerinden istediğinizi tanımlayarak ses ihtiyacınızı karşılayabilirsiniz! İş Avantaj Her Yöne ve Kamuya Özel Her Yöne tarifeleriyle ilgili detay için tıklayınız.
Türk Telekom Ofisleri ve Satış Temsilcileri aracılığı ile hizmete başvurabilirsiniz.
Configuring SIP Trunks / VoIP Providers in 3CX Phone System
On this topic
SIP Trunks / VoIP Providers / VoIP Gateways
Requirements for using a VoIP Provider / SIP Trunk
Requirements for VoIP Gateways
Configuring a VoIP Provider / SIP Trunk
Step 1: Create an Account with a VoIP Provider
Step 2: Conduct the Firewall Test
Step 3: Add the VoIP Provider Account in 3CX Phone System
Step 4: Create an Outbound rule to route calls over the SIP Trunk
Configuring a VoIP Gateway
Step 1: Find the VoIP Gateway and update its firmware
Step 2: Configure the VoIP Gateway in 3CX Phone System
Step 3 - Upload the configuration file to the VoIP Gateway
Step 4: Create an Outbound rule to route calls over the PSTN Gateway
To be able to make outbound calls on the PSTN you will have to configure at least one SIP trunk / VoIP Provider or VoIP Gateway.
VoIP / SIP Trunk providers “host” phone lines and replace the traditional telco lines. VoIP providers can assign local numbers in one or more cities or countries and route these to your phone system. In most cases they also support number porting. Typically SIP trunks are cheaper than traditional PSTN lines. However, be aware that each VoIP call requires bandwidth. 3CX supports both registration based VoIP providers (Trunk logs on with username and password) and IP based trunks (PBX is linked to the Provider based on your public IP).
If you still have traditional PSTN/Phone lines, or prefer to use those, you can connect them to 3CX using VoIP Gateways. A VoIP gateway is a device which converts telephony traffic into data, so that it can be transmitted over a computer network. In this manner PSTN/telephone lines are “converted” to a SIP trunk, allowing you to receive and place calls via the regular telephony network. VoIP Gateways exist for analog lines as well as BRI, PRI/E1 lines and T1 lines. The VoIP Gateway will bundle these lines / ports into a single SIP trunk inside 3CX.
Requirements for using a VoIP Provider / SIP Trunk
- A firewall/router/NAT device that supports STATIC PORT MAPPINGS. Often routers will perform port address translation, which will cause problems such as one way audio, failing inbound calls and so on. It is also highly recommended that you have an FQDN that resolves to a static external IP. If your external IP changes intermittently, inbound calls will fail. See the Firewall & Router Configuration for details to configure your firewall/router/NAT device.
- Adequate Bandwidth. VoIP is real time, so it does place a demand on your Internet connection. As a rule of thumb, each call will consume approximately 30-120 kb per second, depending on which codec you use. The document, Bandwidth Overhead over DSL connections, includes detailed information about bandwidth consumption, including particular codecs bandwidth usage.
- A supported VoIP provider. All supported VoIP providers have been tested for interoperability with 3CX, and are re-tested with each new build. Their configuration templates are also included with 3CX Phone System to allow you to quickly, easily add them correctly. See the list of 3CX Supported SIP Trunk Providers.
Requirements for VoIP Gateways
- A 3CX supported VoIP gateway. Supported gateways have been tested by 3CX and are automatically configured with their correct settings. If using the default configuration, 3CX will also provide first line support of their use with 3CX Phone System. A list of the latest supported gateway hardware, can be found on the Supported VoIP Gateways & ATA's page.
Configuring a VoIP Provider / SIP Trunk
Step 1: Create an Account with a VoIP Provider
First, you need to have an account with a VoIP service provider. 3CX supports most popular SIP based VoIP service providers and we recommend using one that has been tested by 3CX as 3CX includes pre-configured templates for these VoIP providers. Go to https://www.3cx.com/partners/voip-providers/ to see a list of supported providers.
Step 2: Conduct the Firewall Test
3CX will prompt you to conduct a Firewall Test. Frequently, the internet facing firewall sitting between 3CX Phone System and the VoIP provider is not correctly configured or is not able to correctly route VoIP traffic. To check the firewall configuration, it is important to perform a firewall check using the inbuilt firewall checker. To do this:
- In the 3CX Management Console, go to the System Status page.
- In the section “PBX Status” select the “Firewall Check” entry.
- Click “Run.”
- Ensure that the tests for the SIP Port (default port 5060), and the Audio Port range (default ports 9000-9255) pass.
- If the firewall check fails, you must go to your firewall and troubleshoot why the test failed.
Note: 3CX does not provide specific firewall configuration support. Configurations for popular firewalls can be found here.
Step 3: Add the VoIP Provider Account in 3CX Phone System
After you have created the VoIP provider account, you will need to configure the account in 3CX Phone System. To do this:
- In the 3CX Management Console menu, select “SIP Trunks” > “Add SIP Trunk.”
- Select the Country that the VoIP provider operates in.
- Select your VoIP provider from the Provider drop down list. Important: If the provider is not listed, select the “Generic” option in Country drop down menu and then choose between “Generic VoIP Provider,” or “Generic SIP Trunk,” (If using a generic provider we will not be able to guarantee that 3CX will work with this VoIP provider).
- Enter the Main Number assigned to this SIP Trunk. If you just have DIDs and no main number you can select one of the DIDs as the main number. Click “OK.” The SIP Trunk will be created and a new dialog will open.
- Enter a name for this VoIP provider account. The “SIP server hostname or IP” and optional “Outbound Proxy” will be pre-filled. Compare these with the details you have received from your VoIP provider and check that these are indeed correct.
- Specify the “number of simultaneous calls” your provider allows.
- In “Authentication,” specify whether authentication is based on IP or based on Account/Registration. If you selected a template, this will be automatically pre populated and you must leave as is. If IP based, the password will be greyed out, since authentication is linked to your IP. The outbound or inbound only are not applicable in most cases and can be ignored.
- Specify how calls to the main number should be routed. The routing configured here will be for calls matching the main number.
- If you have DID numbers, you will need to specify these in the DIDs tab. Click on the “DIDs” tab and add the DID numbers associated with this account. The DID will be created and linked to the operator extension. You can change this later from the “Inbound Rules” node by adding an inbound rule for the DID and routing to the desired destination.
- In the Caller ID tab, add the caller ID you wish to have appear on outbound calls.
- Click “OK” to save the trunk settings.
Step 4: Create an Outbound rule to route calls over the SIP Trunk
Now you need to create an outbound call rule. To do this:
- Go to the “Outbound Rules” node and press “Add” to create a new rule.
- Decide what calls should be routed over this trunk.
- In the “Make Outbound Calls” section select the trunk you just created.
- Click “OK” to create the outbound rule.
- For more detailed information about creating Outbound rules see this document.
Configuring a VoIP Gateway
Step 1: Find the VoIP Gateway and update its firmware
As a first step you need to connect the gateway and update its firmware. To do this:
- Connect the gateway to the network. Now obtain its IP:
- If using a Beronet, use the “bfdetect” tool. More information about Configuring Beronet BeroFIX.
- If using a Patton, use the “SmartNode” Discovery Tool. More information about Connecting A SmartNode to the Network.
- If using a Welltech, plug in an analog phone into the device and dial #126# on FXS devices. For FXO devices connect your LAN to the device’s WAN port which will acquire an IP address via DHCP.
- Login to the device’s web interface and update the firmware to the latest version.
- Assign a static IP and take note of this IP.
Step 2: Configure the VoIP Gateway in 3CX Phone System
The second step is to create the VoIP gateway in the 3CX Management Console and configure it.
- First of all Update 3CX to download and use the latest Gateway Template.
- In the 3CX Management Console, click the “SIP Trunks” node and then click “Add Gateway.”
- Select the “Brand” and the “model/device.”
- Enter the “Number of Physical PSTN ports on the device.”
- Enter a “Main Trunk number” associated with the gateway. Click “OK.” The Gateway will be added and a new dialog will open.
- Now enter a name for the “Trunk” (VoIP gateway).
- Enter the Hostname or IP of the VoIP Gateway in the “Gateway Hostname or IP” field, and specify the SIP Port on which the gateway is operating. By default this is 5060. Important: Do not change the “SIP User ID” & “Password” fields in the “Authentication” section. The device will be provisioned with these values and use them to register with 3CX.
- Specify how calls to the main number should be routed. The routing configured here will for calls matching the main number.
- If you have DID numbers, you will need to specify these in the DID tab. Click on the “DIDs” tab and add the DID numbers associated with your account. An Inbound Rule will be created for each DID and linked to the operator extension. You can change this later from the Inbound Rules node.
- Click on the “Generate device config” at the top. This will create a file which you must upload to the gateway.
Step 3 - Upload the configuration file to the VoIP Gateway
- Once you have created the VoIP Gateway connection, export the configuration to the device by clicking on the “Generate Device Config” button.
- If using a Berofix, this will open a browser to configure the gateway remotely.
- If using a Patton or a Welltech, the button will download a config file that you must upload to the gateway. Configuration guides for Patton and Welltech.
- Your PSTN lines are now ready for use with 3CX.
Step 4: Create an Outbound rule to route calls over the PSTN Gateway
Now you need to create an outbound call rule that routes calls over the Gateway device. To do this:
- Go to the Outbound Rules node and press “Add” to create a new rule.
- Decide what calls should be routed via this gateway.
- In the “Make Outbound Calls” section select the trunk you just created.
- Click “OK”to create the outbound rule.
- For more detailed information about creating Outbound rules see this document
3CX - SIP Trunking Explained
The trusted old Public Switched Telephone Network (PSTN), with its Analog, ISDN BRI, E1 or t1 lines, is to disappear. Telephony is moving from PSTN to much more modern and flexible SIP trunks.
The big telecom providers are fast phasing out the old PSTN functionality, and are moving customers to IP. And so a SIP trunk and a phone system upgrade in the near future is going to be inevitable.
Verizon will phase out ISDN in the U.S. by 2018. In the UK, ISDN lines are down to less than 3 million lines, from 4.7 million lines in 2007 and the trend is accelerating. By 2017 major telcos such as BT, KPN, France Télécom, Deutsche Telekom and Telecom Italia will no longer offer ISDN lines.
As a result Session Initiation Protocol (SIP) trunking has increased by 62 percent in 2015 from the prior year, driven primarily by North America. A SIP Trunk is usually provided by an internet service provider (ISP). Unlike a PTSN provider, the lines provided are not physical wires, but a service provided over the internet. The SIP Trunk Provider provides phone numbers and lines, usually at better rates than the traditional providers and with more flexibility and shorter contract durations.
This guide explains what SIP trunks are, their advantages and how you can make the move.
What are SIP trunks?
SIP trunks are telephone line trunks delivered over IP using the SIP protocol. Using this standard protocol, telecom service (VoIP) providers connect one or more channels to the customer’s PBX. Phone numbers and DIDs are linked to the SIP trunk. In many cases numbers can be ported to the SIP trunk.
Benefits of SIP trunking
But our farewell to the PSTN brings many benefits. SIP trunks deliver:
- Lower monthly Line & DID Rental – The monthly fee to have a number of lines installed at your office drops significantly with SIP trunks. And DIDs cost a lot less.
- Lower call charges – There are many SIP trunk providers and competition has driven down call charges significantly. Some SIP trunks even come with unlimited calling.
- Better customer service – Provide better customer service by adding more geographical and international numbers. Quickly and easily add numbers to your SIP trunk and terminate them on your IP PBX – you can give customers more options to dial in at a significantly lower cost. Customers can contact you more easily and sales will increase.
- Move offices and keep the same number – SIP trunks are not bound to a location, so it’s easy to move offices without having to change your stationary or inform your customers. There is no longer any need to pay to forward phone calls to the new offices.
- Eliminate VoIP Gateways – SIP trunks will eliminate the need to buy and manage VoIP Gateways. All phone calls come in via IP. No extra conversion often means better quality too.
- Leverage a modern IP PBX – Modern IP PBX / Unified Communications solutions will give customers increased productivity, mobility and boost sales. Connecting an IP PBX to SIP trunks is much easier than via the PSTN.
- Flexibility – It is easy to add channels to your SIP trunk to cope with increased calls. A simple phone call will allow you to add channels, and often this can be done immediately. Compare that to the delay in having additional lines installed and then having to upgrade your old PBX to handle more lines!
- Correct number of channels – With SIP trunks, you can easily choose the correct number of channels that you need. Using ISDN/T1, you often have to choose to add either 15 or 30 lines. This usually means you end up with expensive extra capacity.
Selecting the right SIP trunk provider
The next step is to choose a SIP trunk provider who will supply the necessary SIP trunks. A few factors come into play when making this decision:
- Security – As SIP trunks are exposed to the Internet, it is very important that the SIP trunk has a well secured network and an anti fraud system in place. The anti fraud system must monitor the system and provide protection against call fraud.
- Own network – Does the SIP trunk provider run its own network or is it a rebranded service? There are quite a few providers out there reselling SIP trunks from other providers. Select a provider who has control over their service and network.
- Competitive Cost – Costs vary widely between services. Some vendors will overcharge for SIP trunks. Look for competitive rates, but ensure that you are getting business quality SIP trunk service. For example, telecom providers will provide a cheaper quality to Internet call shops. Be cost conscious, but expect to pay a bit more for business class service.
- Number Porting – Can the provider port your phone numbers? Ensure that you choose a provider who can port all the existing numbers – not all providers are able to do this for all regions.
Upgrading Internet connectivity
Once you have selected your SIP trunk provider, consider a dedicated Internet line for the SIP trunk. Most firewalls are able to handle multiple WAN connections, and, considering the low cost of an Internet line in most places, a separate VoIP connection will be the most reliable way to ensure the quality of your VoIP calls.
However, some SIP trunk providers bundle their service with a dedicated Internet line. This keeps your voice traffic separate from your data traffic. Much will depend on the cost and your network infrastructure. Check that your firewall is up-to-date and will be capable of handling VoIP traffic correctly.
Upgrading the PBX to an IP PBX
Chances are that the trusted old PSTN lines are connected into another trusted old device, the hardware-based PBX. This device its inflexible, difficult to manage and often expensive to maintain. Technically it is possible to buy a gateway that allows the old PBX to talk to the SIP trunks. But why not upgrade to a modern IP PBX and leverage the flexibility and modern features IP telephony can bring. This allows you to take advantage of the cost savings, easy management, and productivity increases with full-scale Unified Communications that an IP PBX offers. You can choose from a hosted PBX, an appliance PBX, or a software-based PBX.
This whitepaper explains the benefits of having a software based phone system. Why it makes sense to move away from proprietary solutions such as Nortel, Mitel, Avaya, Alcatel and Siemens, and from appliance based solutions that attempt to hide the underlying complexity, go straight to a REAL software based solution, running on a mainstream, commercially supported operating system.
Sip Trunking Providers - voip-info.orgThis page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.
Country specific pages:
What Is SIP Trunking?Traditional phone calls go over phone lines, but with the rise of the internet, Voice Over IP communication systems give you much more functionality to receive a high volume of calls. This is called SIP (session initiated protocol) trunking. These services are offered by the majority of VoIP providers and can be tailor made to suit the needs of a small business all the way up to a large corporation with a gigantic workforce and everything in between.
One of the main advantages of going with SIP trunking is that it will integrate everything from data, the internet, your video, and your voice services all into a single line. The term "trunking" is a bit dated and goes back to a time when traditional telephone services were used and it described how a wide variety of telephone users could share a much smaller pool of communication paths, thus making it a much more efficient option for businesses than getting a separate phone line for each worker. That was in the past, though, as today's SIP trunking capabilities will even allow all calls to be considered local calls and won't even require the business to have a single phone line at all.
The Benefits of Using SIP Trunking ServicesChoosing to use a good SIP trunking service will significantly level the playing field for smaller business to compete with large corporations and appear much more professional and bigger than they actually are. For larger business, one of the main advantages is being able to much more easily maintain that local presence. Regardless of the size of the business itself, SIP trunking's biggest benefit overall is simply the reduction in cost to the company's overall expenses. This is particularly true for SIP VoIP services that are specifically crafted for a business in particular.
- It offers very low cost calling.
- It's much easier to scale than other options, making it very future proof.
- SIP trunking makes moving a breeze, as you can easily take your phone number with you while moving offices or even while traveling.
- Network outages are much less impactful, as incoming calls can easily be routed to other locations.
- It's ideal for any sized business with at least 25 physical phones.
- It's a fantastic choice for any business that has an international location.
- It offers the same sort of emergency usages and will even offer enhanced 911 dialing options.
How SIP Trunking Can Take Your Business To The Next LevelIt used to be that you had to have a PRI (Primary Rate Interface) in order to have video, voice, and data all on a single line. This was a physical device that had to be purchased separately but along with your chosen phone provider's calling plans. This is no longer the case with SIP, as it allows your provider to set up a proxy server (commonly referred to as a SIP proxy), which will provide the outgoing, incoming, and PBX calls for you. Everything is located off site and handled by a professional IT team or contractor through your provider, meaning it saves you a ton of time and money on the technical support side of the equation.
Being temporarily shut down means simply having the SIP reroute calls to a different office and also allows you to much more easily outsource to international locations, such as with customer service centers.
All of these benefits are much better than what traditional PSTN (public switched telephone network) have been able to provide through their services and will give your business just what it needs to take it to the next level.
A VoIP is basically a computer network that is adapted to deal with voice calls. Therefore, like any other computer network, it can be hacked. There are a lot of factors that increase the vulnerability of the business VoIP system. One such factor is the connectivity of the VoIP system. Here, ‘connectivity’ refers to the way the network is connected. If the calls are routed to you via the public internet, then the risks are significantly higher. If the VoIP provider is offering you a directly managed IP line straight to your premises and all the calls are routed to you using that line, then the risks are much lower.
Security Tips that You Should FollowThere are a few security measures you should take to make your VoIP network less vulnerable to attacks. Some of the measures that you can take includeUse the Latest in Security FeaturesA number of coding technologies are used to encrypt the data communicated through a VoIP system. You should check with your service provider about the kind of security features that they offer.
IP White listingIP white listing is the process which would allow your network to accept calls from only the trusted IP addresses. However, you would need to have a static IP address for all your destinations. If your business is located in the US and you cater to mainly US customers, then you can block large chunks of IP addresses from outside of the US.
Commercial SIP trunk Providers1CentVoIP is a premier, carrier-grade SIP trunking provider tending primarily to predictive dialer clients. We strongly believe in simple business concepts, so we tailored our SIP solution the same way. Located in Los Angeles, CA, we provide short duration and conversational SIP trunks, hosted ViCiDial and PBX server, flat rate and rate deck billing options.Call us today at 1-8555-1CVOIPSIP Trunking for Asterisk & every SIP Enabled Telephone System. Over 500,000 DID's available in 24,500 rate centers. You can activate and setup service in minutes.• No contract and low rates.• Toll Free DIDs starting at $0.50• Wholesale calling rates and TIER1 quality calls.• Free SMS, Free Support, Free Remote Assistance.• Free LNP on Qualified Project Ports.• Full e911 Compliance• TDM Enterprise quality.• Live customer service, 24/hr ticketing system. Qualified SIP & Asterisk team ready to assist you.• No per channel fees.• FREE API for your website. Automate and increase customer base.• Free Setup on Dedicated Servers & Virtual Servers in our Data Center, if needed.
Introducing: The Next Generation GUI to Asterisk QuBe PBX.All professional modules are provided for free and are included in the installation. No third party requirements on any install modules.Try us out today! Call now: 877-686-4787 or visit us: Asterisk SIP Trunking – US1Call,Inc 1Call understands the callcenter and SMB business demands and seek to under promise and over deliver. Our service and our rates will allow you to seamlessly communicate with your market. Feel free to call us anytime 1-212-845-9704 or email [email protected] LLC is offering A-Z VoIP SIP trunking services to Carriers, CallCenters, Offices( SMBs & SMEs). Hassle free account setup and great tech support. We can provide consulting in setting up your organization's VoIP Infrastructure. A1Routes is your one stop shop for all your VoIP needs. A1Routes provides you with simple and easy to use web portal to setup your SIP trunk, manage your account balance and access CDRs. Need a SIP trunk ? Call Now 1-347-809-3866 or mail us at [email protected] Tandem Inc | Access Tandem Inc provides Tier 1 Dialer, RVM, Premium Dialer, Conversational VoIP Origination / Termination / Toll Free Termination (@.001-paid). Immediate access to 75 carriers, 24x7 customer portal, free test. Ring Less Voicemail Drops and white label RVM.http://[email protected]: john.mcintire3ALTOTELECOM Unlimited Channels SIP Trunk ProviderAltoTelecom is a SIP Trunking Provider VoIP company for Call Centers, hotels, small and large business using a PBX or an Asterisk VoIP based predictive dialer such as Vicidial or Goautodial for telemarketing sales, rates under 1 cent per minute to USA, Canada and UK www.altotelecom.com Online Chat Support AJ-TEL Communications Network Group LLC is offering Wholesale Origination / Termination / Free Toll Free Termination US / MEXICO - We provide high quality, dependable access to over 3100 rate centers and instant access to over 5,700,000 DIDs including T38 and local number portability. Wholesale SIP
--Alcazar Networks - Wholesale Services Over 3,100 DID rate centers. Per minute pricing as low as $0.0005/minute. Per channel pricing as low as $2.00/channel. DIDs as low as $0.10/each. A-Z termination. Over 1,200,000 numbers in DID inventory.Amivox free your phone - Lower your communication cost VoIP provider for both consumers and businesses. Offer's free SIP account. Prepaid and very good rates for network termination with premium quality ( Amivox-Out) . Support for iPhone, Android and Blackberry. Shared balance for multiple users. Calling Amivox to Amivox is free - Sign up for free and try out the service.Anveo offers phone numbers from over 48 countries with instant activation. Anveo's Voice 2.0 Communication and Collaboration Suite with powerful Visual Call Flow technology allows you to visually configure call handling and call termination options for your phone number. Anveo provides FREE SIP trunking and it is one of many termination options available.babyTEL offers SIP Trunking, T.38 SIP Trunking, Electronic Fax and Hosted PBX services across Canada and the U.S.BellVoz offers International and Domestic Long Distance Services with VoIP technology, helping business and consumers to reduce monthly telephony expenses.Best VoIP USA BestVoIPUSA.com offers SIP trunking to private and commercial operators of Asterisk PBX switches. BestVoIPUSA.com also provides all of the well known features of SIP even if you are connecting directly to SIP compliant devices, handsets or servers.Blu-com is an independent full service provider in the field of VoIP and data services.
Box Internet Services offers SIP trunking to private and commercial operators of Asterisk PBX switches. Boxis.net also provides all of the well known features of SIP even if you are connecting directly to SIP compliant devices or servers.Brisnorth Communications Australia Brisnorth.com.au provides SIP trunks, VoIP and SIP Server hardware to Businesses Australia-wide. Carrier-Grade reliable SIP/VoIP services at very cost-effective rates. We can work with your current hardware/phones or upgrade you. We have Plans to suit all budgets and sizes of Business. Contracts and Bundles are optional (Customers are free to go un-contacted and un-bundled) email [email protected] or call 07 3623 0800BroadTone NetworksSIP Trunking service is the most reliable business class IP trunking service for business. If your business needs a reliable trunking solution, backed by serious support, BroadTone Networks is the right choice. Call Today for a free quote. 212-937-8835.Broadvox provides SIP Trunking solutions for businesses with existing IP PBX installations as well as those that are seeking to establish a platform for Voice over Internet Protocol (VoIP) communications over their existing legacy TDM system.BTCVOIP provides SIP and IAX2 termination, caller id, DIDs, no monthly fees, no contracts, simultaneous calling, free calls between customers, all sip and iax2 devices supported,Android client support, link 2 phones together using web callback, you can pay in bitcoins, dwolla, western union, okpay, liberty reserve, also on request encrypted voip using iax2, codecs g711, g729, g723, gsm, and ilbc, you ata will not be locked.btTelco - Cloud PBX provides SIP trunking with CLI, DIDs and hosted IVRs with natural and eloquent Text-to-Speech support voices in more than 15 different languages. Outbound Text-to-Speech voice messages and inbound/outbound SMS and USSD are also provided. Included in the service there's URL callback implementations for each event type and a easy-to-use REST API that will allow the customers to extend the available features or develop new ones.CallForwarding - Be present anywhere in the world with toll free forwarding services from CallForwarding.com.- Interactive Voice Response (IVR), Call Forwarding Services, Automatic Call DistributorCytranetCytranet is one of the largest business VOIP providers in the Southeast, and operates a nationwide network.
- SIP, PRI, Analog, Hosted
- Custom solutions
- Bring your own equipment or use ours
- Fast, friendly support 24/7
- Unlimited calling anywhere in the US
- Keep your current phone numbers
- Call centers
- VoIP providers
- SMB's and enterprises
- Independent software vendors
- Calling card operators.
Conexiant Quality Sip Trunking Provider – direct connections to CLECs for best quality, inc. Level3.
- Global Level 3 footprint Denver, LA, Hong Kong, China and Frankfurt, Germany
- DIDs, Toll Free, E911 service, T.38,VoIP fax, quality termination and dedicated VoIP servers
- DIDs per-channel or per-minute
- Unlimited Local & Long Distance SIP Trunks
- High-Volume Calling
- Use with any PBX
- Fast, friendly support 24/7
- Keep your current phone numbers
- Free phone numbers included
- Works with any SIP based PBX like Asterisk, Freeswitch, Free PBX, Cisco, Opensips, Kamailio, VoIP Switch
- No Commitment, Many products to fit your budget.
- Unlimited calling on SIP Trunks
- Largest Single Tier SIP Trunking with unlimited minutes anywhere.
- DIDforSale now also provides Canada DID's.
- DIDforSale now also provides UK DID's.
- DIDforSale now also offers unlimited inbound/outbound SIP Trunks for businesses.
- Over 1,000,000 DIDs in stock, worldwide. 100% coverage in many countries
- Send 100 or 10,000 SIP channels per account, no extra charge. Serving call centers in the UK/Asia/US/Australia
- Premium SIP termination ONLY. No greyroutes, just ONE single ratelist, carrier direct.
- TDM direct to many LECs. Premium quality - send your CLI, 99% success
- SIP trunk provider for your IPPBX: Asterisk, FreePBX, Elastix and 3CX friendly
- Geo-diverse voice network: Frankfurt, US east/west +Canada, UK, Australia, Singapore, South Africa +Paris/Amsterdam/Stockholm
- sub-1ms latency to several exchanges. G711u/a, GSM, G729 and G722 HD Voice supported
- Full HTTPS, Geotrust verified, audited. Get started with $5 Paypal payment, send SIP calls in minutes.
eBestPhone Communications LLC is a competitive VoIP service provider for international call termination. We can provide you with a low-cost solution to all international destinations via our SIP trunking network and can offer hosted prepaid calling card services on dedicated servers. Our network is fully redundant with diverse fiber and microwave access. We are very competitive to destinations in Cuba, Mexico, Honduras, Guatemala, Ecuador and many more. Latin American countries are our specialty. We offer basic Mexico rates for fixed and wireless at $0.0296/min and $0.1039/min, respectively. Email [email protected] for more details.ePhone VoIP Service Retail and Wholesale VoIP Telephony Provider located in Greece with Direct Termination Routes. ePhone can support large number of concurrent calls and call initiations per second. ePhone aslo provide Greek DIDs, Fax2Email Services, Hosted IVR, Bulk Faxing, Voicemail Broadcasting, Hosted PBX, Calling Cards and SBC Services.
Fiber Vision Networks Pty Ltd Fiber Vision Networks delivers VoIP and SIP trunk services throughout Australia, delivering high quality telephony services without any call compression.Instant Setup. No Setup fee’s, Commitments or Terms. Full Termination and Origination DID’s Available in all US Area codes. Rates as low as $0.003 per minute. Live US Support. Call 24/7 to 800-834-8999 [email protected] Inc. As the world’s first pure SIP certificated carrier, Flowroute delivers advanced calling and messaging communications to answer the needs of developers, SaaS service providers, and enterprises. This certification enables Flowroute to give developers direct access to telecommunication resources such as phone numbers, inbound and outbound calling, messaging and advanced signaling, to deliver carrier-grade voice and messaging experiences. Through its patent-pending technology, Flowroute provides developer’s unparalleled performance, transparency and control to add voice and messaging into their apps and services to create unique user experiences.Fractel Origination and LNP in more than 10,000 North American rate centers including Canada, Alaska, Hawaii, and Puerto Rico. Free CNAM on all DIDs. Full service Toll Free with no RespOrg fees. Aggressive pricing on termination to 500,000 US and A-Z international routes with full CLI. Data services (411, E911, LRN and CNAM dips). Top quality, fully redundant network with 100% uptime guarantee. Full control of all services with easy to use portal. Comprehensive plans or ala carte services. Monthly contract. Free test accounts.FreelyCall Competitive low rate termination to most countries, retail and bulk DIDs to 80 countries, Direct IP (guaranteed CLID) to 25 countries, Local Number Portability to 10 countries, special prices for quantity orders.
Geils Communications - US based IP carrier serving global customers with custom services.Momentum SIP Trunking is a business-class solution designed to deliver affordable Voice over IP service with advanced telephony features to an existing PBX system.GLSIP - Canada - BURSTABLE SIP Trunking Services from the experts at Group of Gold Line (GL).
- Enjoy the flexibility of a burstable GLSIP trunking solution to support your dynamic business environment
- Highly reliable, highly scalable and secure SIP trunking services for mission critical enterprise telephony needs
- Lower your cost of fixed trunking requirements
- Global Point of Presence (POP’s)
- Most competitive Toll Free and Long Distance charges
- Replace/Eliminate expensive PRIs and analog facilities
- SIP expertise and technical support (24/7)
- Simplified contracts with no restrictive conditions
- Global Virtual Number presence (Global DID)
- No charge, no obligation proof of concept trial. Try it, Like it, then Buy it.
- Our technical expertise extends “beyond the demarc”, unlike our competitors
ITG Telecommunications - Malaysia Corporate and Wholesale SIP Trunking, A-Z Termination, Hosted PBX, Local DID numbers in Malaysia. Fast technical response time and superb voice call quality.Junction Connections Wholesale & Retail SIP Trunking. Origination and A-Z termination. Unlimited inbound USA offered. Great for calling card providers. Pay per minute also available including E911 & CNAM.Kesher Communications (UK) - Business VoIP Telecoms provider for small to medium sized businesses. Free SIP trunks for Asterisk and 3CX and lower call costs.
LIGA Telecom is an established Wholesale Aggregator of Retail Voice Traffic from Cable, Mobile and Retail Operators focused on recently deregulated and developing markets, normally considered risky. We are a PREMIUM international telecommunication carrier specializing in delivering ONLY high quality and stable VoIP Termination service at competitive wholesale rates to Retail Providers and Wholesale Carriers, since 2010.
IP PBX Support Inc. - Business grade SIP trunking services. California based.
Incorpus TeleNetworks One stop solution for all your voip needs.Dialer or call center routes like USA flat, USA npanxx, uk , Australia and many more, call center plans, wholesale or retail routes like India Cli/ncli, BD Cli/ncli, Singapore Cli/ncli and A-Z Cli/ncli , websites, hosting DIDs, Tollfree, SIP trunking, White label reseller, Private label resellers. Class5 sofswitches, Class 6 softswitches, Carriergrade softswitches all with one time installation and monthly rental plans available. email at [email protected] for more information. Logic Voice SIP Trunking service is an award-winning converged voice over IP solution that allows you to capitalize on your investment in data and voice communications and get the most from your IP-PBX today.
Lunafon SIP Trunking with Reseller opportunities; low rates and excellent quality. Cloud PBX services with Swiss DIDs.Magnetic North is an international provider of a wide range of online VoIP services including SIP trunking, Cloud PBX, Cloud Contact Centre and Cloud Recording. Visit our website for more information.MasergyMasergy Intelligent SIP trunking delivers more than just call origination and termination. It delivers productivity-enhancing features to your desk phones and beyond. Their application-oriented platform that delivers true work from anywhere connectivity from the cloud. By overlaying Direct Inward Numbers with our enhanced features, a premise-based PBX user can benefit from modern productivity tools without added hardware costs.Metric Infinity SolutionsAustin based small to mid-size business IT services consulting. Offering SIP Trunks and hosted virtual phone servers using Asterisk and FreePBX. Our SIP services are billed per minute, allowing unlimited concurrent call paths. Sign up for a trunk and get started within minutes! Some of the lowest rates in the country. United States dialing at $0.0098 per minute. Offering vFax and SMS services too. MOBEX offers SIP trunks, Hosted PBX and Virtual PBX providing the best in office phone service. Mobex, the Nation's local business phone company, provides small, mid-size, Mobile Offices and Call Center businesses a sophisticated, yet inexpensive telecom experience with customizable call handling features to help better connect with customers and improve internal procedures. Mobex is also a chosen VoIP partner with Verizon FiOS. To learn more now how we can improve your company's communications, be sure to visit us at Business Caliber Telecommunications or to contact us please visit Contact MobexAdvanced phone solutions, tailored for your business. Mobex is a professional, reliable phone service that improves your company’s image and productivity. MultiTELQuality A-Z termination SIP trunking with no call setup fees. Pay for what you use. Terminate calls from your authorized IP addresses or via SIP registration. Benefit from additional services - Hosted PBX and International DIDs (over 100 countries) . Points of presence in both US and Europe. Net2Phone Net2Phone offers high quality, low cost SIP Trunking and Business VoIP Solutions - no charge per channel, no contracts, pay only for the minutes you use. Net-Voice Telecom NVT is a Leader in IPPBX and Sip Trunking with Office Located in NY and FL we have extremely competitive Packages for Small to Enterprise class BusinessesNeuwise Communications provides SIP Trunking and SIP Peering Solutions for businesses/enterprise and retail subscribers, including Wholesale (A-Z and desired specific prefixes) termination services world wide using 4 POPs (Dallas, Singapore, UK and Netherlands).Offering A to Z VoIP solutions including DIDs, Class5 full telephony services and features and HD Voice (recent Pilot) nexVortex nexVortex is a premier Internet Telephony Service Provider (ITSP) offering SIP Trunking services to business customers. All of the inherent benefits of SIP are realized with nexVortex, as well as the support of a financially stable, growing company dedicated to exceeding its customers expectations each and every day. Optivon - Helps to maximize your operations and gain competitive advantage and business growth.through SIP Trunking and cloud telecommunication services.USA Services Include: SIP Trunking Virtual PBX Cloud ACDCall 1-844-OPTIVON or visit www.Optivon.com for a free quote. PHC Tailored Telecom PHC is a Dutch Telecom oriented company specialized in communication solutions for companies. VoIP, WhatsApp for business and integrated telephone services are common solutions provided by PHC.PortaSwitch is a single software platform which delivers private branch exchange (PBX) functionality as a service and can be successfully used for SIP trunking business supporting:
- Inbound and outbound calling
- Toll-free and Direct Inward Dialing (DID) numbers
- All current regulations including E911 and CALEA
- SIP over TCP, particularly for customers using Microsoft OCS
- National directory listing
- Multilingual online real time billing and account management
- Unlimited Extensions
Introducing:The Next Generation GUI to Asterisk QuBe PBX.All professional modules are provided for free and are included in the installation. No third party requirements on any install modules.Try us out today! Call now: 877-686-4787 or visit us: QuestBlue Systems Inc Get the best rates on sip trunks from a global provider with the best call clarity and highest reliability
- Setup in less than 24hrs
- Unlimited local and long distance for one rate
- Amazing International calling rates
- Local, Toll-free, vanity, International and virtual phone numbers
- Thousands of rate centers all over the world
- Split second failover you can configure to your needs in real time
- 24x7x365 access to make changes to your service
- Live support
- SIP trunking
- Hosted IP PBX services
- Class 5 IP Centrex solutions
--Sip Trunk to Viber Gateway Sip to Viber Gateway. Viber is used as a mobile sip client or IP phone. The integration of IP telephony and Viber app. For SIP providers easy setup on the client side. One Viber number = $ 1 / month. Unlimited incoming calls.SIPTRUNK.com - The simple, easy and profitable way to resell SIP trunking services.With SIPTRUNK.com, account creation to live service happens in less than 10 minutes! The SIPTRUNK.com white label SIP trunking platform is designed for hardware-centric businesses interested in developing a monthly recurring revenue stream from selling SIP trunking services. SIPTRUNK.com is brandable, handles all end-user billing and complex telecom taxation collections, can be configured for multiple partner levels and provides transparent commission reporting all backed by service and support personnel with decades of telephony industry experience.China SIP Trunking (CTS):A Leading Business VOIP Solutions Provider in China. Hosted PBX in China,SIP Trunking in China(Shanghai Number),VOIP Consulting, VOIP system Maintenance, Lync Integration.
Star Communication offers SIP Trunking that is reliable, redundant and built for call quality. If you need to urgently get out 1 million calls in an hour, we give you access to 45,000+ channels at your fingertips. Typically we are able to offer better rates and assist our customers in choosing the right solution. We have helped SMBs, Large Call Centers, VoIP carriers, CLECs and we look forward to assisting you! Visit our site below for more information and pricing.To create a FREE test account, simply complete our online form here
Full-time NOC supportTier 1 direct interconnectsHigh CapacityHigh CPSFree Testing
Deal with the Security of Business VoIP Networks with Switch3voipSecurity Concerns with Business VoIP Smart Voice Network offering service to Carriers and Call Centers with unique ability to cap rates and manage sub accounts. VoIP Carrier with US & International Canada $.009 UK, France 0.0082,Mexico $0.009, China 0.0088 for domestic $.008 USA Termination and Origination, offers termination & origination for Mexico $0.009 per minute and DID's for $1.00__ per month all with unlimited channels / concurrent calls. Competitive High Quality NPA-NXX rate deck, International Carrier rates, and DID channel pricing. For calling cards, mobile dialer, Efax solution, Bulk SMS and white label for the best web conference we will host it for you and payment solutionCall us at 760-517-6724.Email us: [email protected]: william.shihata1Spectrum Business - Spectrum Business provides SIP Trunks and high-speed internet for businesses. Incoming Phone Numbers, SIP Phone, PIN Protected Phones, Cloud PBX, Hosted PBX
We offer unbeatable prices on unmetered DIDs, Toll-free, shared and metered numbers in 60+ countries, and outbound call rates as well as Cloud and Hosted PBX solutions with data centers in USA, Germany, UK, Japan, Singapore.Sonic Communication is an wholesale VoIP Provider offering cost-effective and best quality wholesale VoIP termination. We offer high quality and low cost Wholesale VoIP Termination.
- We have both call center routes and long duration direct routes. http://soniccommunication.com
- USA, Canada, Dialer, Call Center, Business, Tier 1 Direct
- Unlimited local & long distance calls within 48 US states
If you are looking for a reliable VoIP provider with high quality service, competitive rates innovative services, and flexibility, you have come to the right place at the right time!
Telecom Call Center is a fast growing telecommunication service provider in Europe.
We offer more than 90% CLI Routes to carriers, Small Businesses, Call Centers (Predictive Dailer), Callshops, Pc to Phone, Mobile Calls App provider like FreePP Calling Card providers, Resellers and other VOIP service providers.
Starting with us is fast and simple.Write us an email with a short introduction and your requirements and contact details.We will schedule an interconnection and test account.Make a prepayment and benefit from Premium routes from world’s leading PTTs.
88- Stratics Networksis offering A-Z VoIP SIP trunking services to Carriers, CallCenters, Offices( SMBs & SMEs). Hassle free account setup and great tech support. We can provide consulting in setting up your organization's VoIP Infrastructure. Stratics Networks is also the largest provider of outbound IVR and inventors of Ringless Voicemail Drops. Stratics Networks is your one stop shop for all your VoIP needs. A1Routes provides you with simple and easy to use web portal to setup your SIP trunk, manage your account balance and access CDRs.Telnyx - Next-Generation Sip Trunk ProviderTelnyx is a provider of wholesale VoIP services. Our goal is to democratize the Public Switch Telephone Network and transform the way you communicate every day. We do this by enabling you to "Be Your Own Carrier®" through our innovative Mission Control self-service portal and RESTful API, supported by our private network to ensure quality and security.
We are a leading SIP trunking provider because we give our clients the power to purchase services a la carte, giving them the flexibility to only pay for what they use. Here are just some of the features that you can take advantage of in our Mission Control platform:
• Origination and Termination services• Instant number search and provisioning from over 15,000 rate centers. International and toll-frees also available• Anchorsite™ - gives you the power to decide which point of presence (PoP) you'd like your media to be anchored. Current presence in Ch2, DC2, SV1, LD5 HK1 with plans to add 12 PoPs in 2017• CNAM, call forwarding and e911 services• Network security with TLS and SRTP/ZRTP encryption• Real-time reporting for CDR requests, monthly charges and usage reports• 24/7 support for any of your technical needsAnd with partners like 3CX, Microsoft Skype for Business, Asterisk, Cisco and Avaya, you'll be sure that you'll be working with a trusted name in the industry.T38 Fax Services T38faxservices by babyTEL is T 38 Fax over IP that works. Get a T.38 SIP Trunk that provides affordable, reliable Fax over IP service. Pricing starts at $8/month, with the first 30 days free.
Telzio Small business phone system with unlimited SIP users from $1/month
- Ubity is the pioneer for hosted communications in Quebec.
- Ubity also provide SIP Trunks with UB-SIP and DIDs all around Quebec, Canada, North America et the rest of the world.
- Ubity's UB-VoIP system is an Over The Top (OTT) solution and do not require IP-PBX equipment or maintenance.
- Thanks to UB-UC, Ubity's Unified Communications as a Service (UCaaS) solution, all modern channels are in the same application
- With UB-API you can manage your integration between our application andyour software such as CRM / ERP (Saleforce, Netsuite...)
• High-end hosted IP telephony service• Web portal that allows you to manage your telephony service• BYOD with Multiplatform PC and mobile software (Windows, Mac, Linux, Android, iOS, Blackberry, Windows Phone...)• Secure chat and file transfer between employees• HD quality voice and video chat• Call recording• …and more
Reach out for more information
Unlimited SIP Trunk.comUnlimitedSIPtrunk.com offers single-source custom SIP trunks with unlimited calling. Call 800-861-1183.
- Compatibility with 200+ PBXs and devices
- 99.999% reliability
- Keep your local numbers or get new ones
- Free eFax
- Unlimited calling anywhere in the US
- Premium calling features such as Simulring, Robocall Blocking.
Spout Communications We provide high-quality Business Wholesale VoIP Termination, Canada/USA/Toll Free/International DID Origination, Toll-Free, Number Portability, dial back, affiliate program, consulting (remote/on-site), trunking services, at a very affordable cost. Experience Tier 1 carrier-class quality, and significant savings.
2 - Tel2 - Feature-rich VoIP provider with FREE signup and UK DIDs. Offer a host of services including call recording, web and video conferencing and collaboration services, faxmail (+ T.38 passthru), locate me, Smartphone and Desktop Apps and more. We offer all UK landline and tollfree numbers. Suitable for all users from residential, business through to large call centres. Wholesale and reseller programs and a white label 'Telco in the Cloud' product available. Lowest Rates. Save with calling bundle rates of 0.6p for landlines, 2p for Mobiles. 40+ countries at 1p/min. Build your own Telco in the cloud under your own branding and set your own rates and create your own calling plans and bundles using our fully automated web portals.Tekara VoIP - SIP trunking providers offering peering and termination for all UK and Internation destinations. UK Provider of VoIP and Numbering Services and cheap UK and Internation Calls. Business and Home VoIP Solutions.
Continent Telecom – offers international DID numbers in 90 countries, Toll-free, Fax, SMS DIDs.
Virtual Voice Connect Number 1 for business solutions in the UK, we are a hosted VoIP telecom provider for UK Businesses. Virtual Voice Connect provides VoIP DID numbers, number porting, IP Trunking, and guide you towards better trunking security and connectivity, as well the UK's cheapest SIP Trunks for just £2.00, available with a choice of DDI's. Call us on +44 (0) 844 310 4635 to find out how we can help your business today.SIP Trunking for businesses offering service to PBX and wholesalers. VoIP Carrier with UK , USA & International Termination and Origination. Very low prices with good quality for termination, and very good DID global channel pricing. For calling cards, business grade pbx , Efax solution, Bulk SMS and white label for the cloud hosted pbx solution Call us at +1 518 2160863 (USA) or +353 148 51863 (Ireland).You can create a FREE demo account for testing FREE demo account click here CloudSpan by VoIP Supply: Get a Free Consultation!Let our VoIP Solution Specialists pair you with the ideal service provider!Already well-known as the trusted one-stop-shop for your name-brand VoIP hardware needs, VoIP Supply now provides you with a single place to shop various VoIP service providers - finding you the perfect match for your business's needs.
Multiple Service Providers
We have partnered with the most trusted service providers that line up with our core values and customer satisfaction. We will present you with numerous proposals and guide you towards the perfect solution.
Our VoIP Solutions Specialists are trained by multiple service providers, to choose the best solution for you. We save you time and let you focus on your businesses needs.
By leveraging our relationships with our trusted providers you will not need to sign any contracts. With our money-back guarantee you can be sure we will choose the best solution for your businesses needs! We want to insure we are choosing the best solution for your businesses needs, that's why we honor a money-back guarantee.
What's the First Step?
You will first need to decide whether you would like to have a provider host your phone system off-site in their building or bring the hardware to your office and take care of it yourself.VoIP Innovations is a Wholesale VoIP Services Provider, interconnected with the industry's top telecommunications carriers. VoIP Innovations provides high quality and low cost VoIP services to ITSPs and resellers through our unique industry leading BackOffice, Titanium III. Titanium III provides you with online access to the industry's largest footprint and number warehouse.
- Largest DID Origination Footprint Available
- DIDs, Toll Free, E911 Service, VoIP fax, SIP Termination.
- Lowest DIDs Usage Rates in the Industry - USA, Canada, and 60+ Countries
- E911, CLI, 411, CNAM, T38, Call Forwarding, and Failover Services
- Have complete choice, automation, and control with the Industry Leading wholesale VoIP BackOffice
- Other features included in the BackOffice include Hosted Billing, Fraud Detection, and an End User Portal
Voicebuy VoIP Provider Wholesale VoIP termination, A-Z SIP trunking, Business Phone Systems, Mobile VoIP Solutions etc.VoipTiger Offers a Free cloud pbx - Sip trunk services - Call center features - Numbers (DIDs) in 53 countries - Free Android/Iphone app - Grandstream IP-phones - Codecs: ILBC/G711/G729/G722
VocalNet VocalNet offers IP origination and termination from all 50 states and around the world. Supported codecs include G.729a and SIP. Since 2001, VocalNet has been offering local, long-distance and toll free numbers all over the world.VoIPPBX.systems The white label, commercial grade IPPBX | VoIPPBX.systems doesn't sell to clients directly. We are a wholesale business that will only work with a strong reseller / dealer network. Be part of the VoIPPBX.systems network if you want to be able to offer a trustworthy solution to your clients.
- Step in at various levels.
- Online Interface, fully hosted solution.
- DIDs in 60+ countries.
- All Class 5 features.
- Tons of features.
- Based on PortaOne.
- Even use our platform to do the billing for you.
White Paper to Find out a Wholesale Sip Provider for your business needsThe white paper will guide you in selecting a wholesale sip provider for your buisness needs and it will discuss the important factors to consider before you make decision Find a best wholesale Voip Provider for your business
X-on SIP Trunking
Highly Flexible Alternative to ISDN
X-on SIP Trunking allows organisations to connect directly into the X-on Network via DSL or Ethernet connection, enabling full PSTN connectivity.
X-on SIP trunks are cheaper per channel than ISDN, offer much more phone number flexibility, are installed quickly and provide reliable voice connectivity ensuring you don’t lose any calls. We can support connections from small PBXs right up to large scale call centres.
Once we've established the required bandwidth and connections, X-on connects to your systems providing end-to-end connectivity with guaranteed call quality, availability and support levels.
SIP Trunking assists your Organisation:
- Number flexibility (e.g. work from Manchester via a London Number)
- Lower call costs with free calls between IP sites
- Multi-site Support, facilitating ISDN Line rationalisation
- Integration with existing systems
- Fully scalable per channel
- Call forwarding at no cost ensuring Business Continuity
- Convergence of Voice and Data via a single access point
http://www.phone2net.com Phone 2 Net offers SIP , IAX2, and h423 trunking services. Get unlimited incoming calls over IP to any of the protocols or Google, MSN, or Yahoo messengers. Guaranteed tier 1 quality numbers only. Best price in the market. visit www.phone2net.com
DIDNumbers.com : Wholesale DID Numbers, Global Coverage for VoIP carriers, Webportal & API
Rightcom NL - Dutch Business VoIP Telecoms provider for small to medium sized businesses. Free SIP trunks for AVAYA, ALCATEL, SIEMENS and NORTEL.
DID World Wide International numbering provider covers over 50 countires. Instant activation, forwarding to regular phones (PSTN), SIP, IAX, h423, GTalk, Messenger and Sype.
GGP Telecom Serving Costa Rica and Latin America markets. Wholesale A-Z termination and, US and Costa Rica origination. Wholesale SMS over IP provider. We beat any competitor's price.
Tpad provides global SIP Trunking or IAX2 Trunking services to businesses around the world, along with high class wholesale VoIP / SIP termination, custom built SIP predictive Diallers, Call Reporting, Call Recording, pure A-Z origination, toll-free, DDI, DID numbers and asterisk telephone systems.
Supanet offers any business SIP Trunking or bespoke IAX2 trunks along with high class wholesale VoIP / SIP / IAX termination, clean origination, unlimited toll-free, DDI, DID, ISDN, SDSL, VDSL, Dark Fibre, FTTC, MPLS, VPN, CPS, bespoke Asterisk VoIP predictive Dialers and Line Rental facilities.
http://www.a3mgroup.com/ A3M Group specializes in all communication options and facilitates channel partners and companies to reap benefits from market opportunities within their core business and customer base. We offer customer centric solutions like Managed VoIP services for resellers, Carrier Interconnect Partitioning, White Labeled Calling Card and Soft Phone services along with Carrier Grade core network elements for established offices to shift their traditional network to NGN
http://www.VirtualPhoneLine.com World's first company to provide a phone number on an IP device in year 2000, continues the tradition to offer SIP and IAX2 trunks, call forwarding to PSTN, MSN, and Google Talk messengers. Try numbers on it free for 25 days, no charge.
VoicePulse for Business SIP Origination & Termination for your PBX. VoicePulse Business provides services to businesses, wholesalers and reseller around the world. Free SIP Trial Account available.
SigmaVoIPSigmaVoIP.com US Based SIP trunk provider.
- 3CX Preferred provider with 100% 3CX Advanced Certified Staff
- Interop certified with Grandstream UCM, Epigy, Digium and many more
- Unmatched support, global presence and extremely competitive rates
- Outbound Domestic Calling from $0.015 to $0.008 ultra low International rates.
SIPRoutes.com The lowest cost SIP Termination option in the market today. As a cloud-based proxy, and integrated into every major US backbone, our service allows our customers to directly access more SIP Termination providers than ever before.
SIPSaver.com SIPSaver offers measurable cost savings featuring free long distance in North America, flexibility to add trunks in minutes and uncompromised quality of service. USA and Canada at only $0.008 /minute.--siptrunkingproviderFirst choice for HIGH AVAILABILITY HOSTED VOIP PBX in the UK
New UK mobile packages now available! 5000 mins £149.99/month (3p/min)
-FREE Setup-FREE HA SIP Trunk Included-Lower Call Charges-Voicemail-Queues-IVR-Call Forwarding-Hunt Groups-UK Mobile 4p Per Minute (main networks)-£2.99 Per Extension-FREE Call Recording-Guaranteed Uptime
http://www.siptrunkingprovider.info--SipTrunks.org Siptrunks.org offers SIP trunks from multiple SIP trunk providers, from SMB PRI replacement to SIP trunks for call centers.
Svanto.net - Tailored Internet telephony solutions Worldwide VoIP provider for residential, wholesales and business. Providing international DID's via high quality SIP Trunks. Easy to connect your Asterisk, freepbx, Elastix, Trixbox, Alcatel-Lucent Pbx, CUCM, Avaya and many more.
Teledynamic Communications Local San Francisco Bay Area provider of SIP trunks, hosted PBX, Digium Switchvox and Xorcom. On-premise installation and service. 510-3422-4200 Option 6.
Teliax provides business phone service through their unique IVY interface. IVY allows you to setup SIP trunks or hosted PBX features...or even both. Features include:
- Ring Groups
- Voice Menus
- Conference rooms
- Voicemail to email
- Fax to email
Tritel BV Affordable, Reliable and Accessible
- Vertiro provides technologies, tools, the skills and the know to create smart solutions that enable companies, employees, customers, partners and other stakeholders to communicate and collaborate effectively. We design and deploy solutions of all sizes, from simple systems for small & medium business, through to converged communications solutions suitable for corporates with thousands of employees.
Voice IT High quality hosted VoIP provider with reliable support.
VoiceStep Telecom provides SIP Origination and Termination to Businesses, Call Centers, Calling Card Operators and other Wholesale Carriers. We have direct contracts with Verizon, Level3, Qwest, etc... We offer Level3's ELS product and have aggressive pricing on Level3's entire footprint. Contact us today at 949.528.3000 Option 1 or [email protected]
VoiperCan Complete solution for routes quality testing for small companies and for big telecoms.
Voys B.V. - We change business telephony. High quality VoIP Trunks and Hosted VoIP accounts with the best service.Voyced | Voyced is a VoIP (telephone) solution provider with the ambition to grow to becoming one of the main pan-European players with our low-cost, commercial grade IPPBX and VoIP services.
We are already present in over a 150+ countries with Standard, Call Center, TollFree and /or UIFN telephone number / DIDs for residential and business users and expanding further. You can make phone calls to anywhere in the world without any extra crossing country border costs as all our clients get the same tariff, no matter where they are located. Quality and flexibility are always most important and now you can have that, wherever you are and for the lowest cost ever. Let us help you and get in touch now!
In the end, it boils down to the fact that we supply any business with an award winning, commercial grade VoIP solution.A telephony solution that would normally only be available to the big corporations and for a far lower price then you would think possible.RingVoz | RingVoz is a VoIP SIP Trunk provider provider offering international and domestic services.MINUTES ANYWHEREOutbound calls to any destination from your house phone, your business, or both, with unlimited plans for USA, Canada, and Puerto Rico.MINUTES ANYWHERE FOR CALL CENTER AND SPECIAL PLANSLarge blocks of minutes at low prices for outbound calling operations; you may include functionalities to manage calls such as intelligent and predictive dialers. If you have a special requirement, we can configure it to you exact needs.ANYWHERE PHONE NUMBERYou can buy a telephone number so that anyone may call you locally in another country and your smartphone or your business phone rings, no matter where you are located. For your business, you have the option of a 1-800 or TOLL FREE, as a courtesy for your clients.SMS BASED WEB MARKETING (WEB-TO-SMS)Your target audience uses text messaging (SMS) to communicate, so it makes sense to complement your marketing campaigns with SMS-based promotions. Our Product / Service TwixText ™ provides precisely that complementarity for your promotional initiatives. VOXGATE offers SIP Origination and Termination to Businesses, Call Centers, Calling Card Operators and other Wholesale Carriers. Attractive rates for Asterisk call termination in EUR. Many payment options available, such as credit cards, PayPal, Skrill etc.http://www.winet.ch Swiss SIP service provider for business and private. Fast and easy installation of VoIP PBX made in Switzerland.
NGNsystems Ltd. provides business class VPBX in Russia. Unlimited rates. We support SIP different CRM systems calls from Skype and etc.
ZONE Limited is a leading ITSP provider in HK offering China / HK / Hong Kong / Singapore / SG DID via SIP trunk or PSTN forward.
IAX2 Trunking Providers
Mountain VOIP Communicationshttp://www.voip-info.org/users/view/mountainvoip| MOUNTAIN VOIP COMMUNICATIONS is the leader in VOIP SIP Trunking services. We offer Unlimited SIP Trunking services for USA and Canada dialing plans, Residential and Commercial phone services with Enterprise features. Each service has web console via which customers can upgrade their service or make a payments.
- Phone lines
- Hosted PBX
- Dedicated PBX
- Unlimited SIP Trunks
- DID numbers
Sipmobilehttp://www.sipmobile.org| VoIP provider for mobile phones. Located in Slovenia. Support for iPhone, Android, Windows phone. Free SIP account. Free audio, video calls, messages between Sipmobile users and hundreds SIP networks. International calls at very attractive prices. Secure communication. Good quality mobile calls using 3G and WiFi networks. Sign up for free and try our service.
INTOhttp://into.nu/ INTO is a VoIP and Cloud provider based in The Netherlands. Feel free to contact us and ask for more information.
Zoom Soft is worldwide SIP Trunking Service Provider and Dedicated Server provider. With 24/7 services we always try to give our best services for client. Our services are given below:
Connecting two IP PBX box using SIP TrunkSuppose you have two IP PBX box one at INDIA and other at US and you want to integrate all the SIP extensions created on both the server with each other so that they can make calls to each other.
IP PBX 1 (India)SIP Extension : 1000, 1001192.168.0.100
IP PBX 2 (United States)SIP Extension : 2000, 2001192.168.0.200
We will have to create an internal SIP Trunk connecting both the PBX box. Follow the below mentioned steps to do the same
Configuring IP PBX for server 192.168.0.100Click on Connectivity and select Add SIP Trunk
Name you Trunk for my case SIP_Trunk_to_US
Configure your Outgoing and incoming settings as shown below. Create a user for incoming calls and one for outgoing which will be used on the US server for authentication purpose.
Configuring IP PBX on 192.168.0.200 (US)Open the SIP Trunk as shown above with the following configuration. Use any Trunk Name as in my case SIP_Trunk_to_IN Configure the Outgoing and incoming settings as shown below. Be sure you use the same user for incoming which you used for outgoing in INDIA SIP Trunk.
Save and apply the configuration. Now we need to configure outbound routes on both the servers
IP PBX 192.168.0.100 (India)Click on Connectivity and select Outbound Route
Configure the route as shown below
The dial plan I have used is X. which means that any extension from IN can call to any extension in US.
IP PBX 2 192.168.0.200 USConfigure Outbound trunk with the following settings
Click on Submit and Apply the configuration on both the servers On the home page of your PBX web console you should be able to see IP Trunk Online
Now you should be able to make calls from 1000 to 2000 i.e from India PBX to US PBX.
Let me know your success and failure.
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