AG Telecom Services

Sip Trunking

SIP TRUNKING

Sip Trunking

SIP Trunking is a cost-effective solution for businesses looking to streamline their communications infrastructure. It enables the transmission of voice and other unified communications services over the Internet, eliminating the need for traditional telephone lines. By adopting SIP Trunking, businesses can take advantage of the scalability and flexibility of IP-based communications while reducing costs associated with hardware and maintenance.

  • Cost Effective
  • International DIDs
  • Industry Standard Security
  • Relaible and Flexible
  • Local DIDs
  • Toll-Free DIDs
  • Highly Scalable
  • Canadian Support

Fully Compatible & Supported

With AG Telecom and Networking’s SIP Trunking solution, businesses can easily add or remove channels as their needs change and only pay for what they use. This allows for greater control over costs, as well as the ability to scale up or down as required. Additionally, our SIP Trunking solution offers a range of advanced features, such as call routing, caller ID, and automatic failover, ensuring that your business is always reachable and your communications are secure.

  • AG Telecom
  • Cisco
  • Shoretel
  • Nortel
  • 3CX
  • Panasonic
  • Mitel
  • Astersik

Knowledgeable Support

At AG Telecom and Networking, we understand the importance of reliable and efficient communication for your business. Our SIP Trunking solution is backed by our team of experienced professionals who provide 24/7 support and monitoring, ensuring that any issues are resolved promptly. With our solution, you can be confident in the reliability and quality of your communications, allowing you to focus on your business goals.

Why Choose Us

PBX Integration

Our SIP trunking service seamlessly integrates with your existing PBX system.

SIP Compatible PBX Support

We provide support for a wide range of SIP-compatible PBX systems.

High-Quality Codec

Our high-quality codec ensures crystal clear voice quality for all your calls.

Flat Monthly Rate

We offer a flat monthly rate with no hidden fees or charges.

Easy Setup

Our SIP trunking service is easy to set up, so you can start using it right away.

Secure Connection

We use secure connections to ensure the privacy and security of your calls.

HD Voice Quality

Our SIP trunking service supports HD voice quality for a superior calling experience.

Amazing Service

We provide exceptional customer service and support to all our clients.

SIP Trunking Pricing Plans

SIP Trunking

SIP Pay as you go

£ 6.95 /Month

  • No Setup Fees
  • Incoming Calls
    0.000/minute
  • Outgoing UK Calls
    0.02/minute
  • Increment:
    6 seconds
  • # Channels:
    One (1)
  • Voice Mail:
    Free
  • Local DID:
    1 Free


signup now

SIP Trunking

SIP With Bundle 1000

£ 10.95 /Month

  • No Setup Fees
  • Free UK Calls
    1000 Minutes
  • Outgoing UK Calls
    0.02/minute
  • Increment:
    6 seconds
  • # Channels:
    One (1)
  • Voice Mail:
    Free
  • Local DID:
    1 Free


signup now

SIP Trunking

SIP With Bundle 1500

£ 15.95 /Month

  • No Setup Fees
  • Free UK Calls
    1500 Minutes
  • Outgoing UK
    0.02/minute
  • Increment:
    6 seconds
  • # Channels:
    One (1)
  • Voice Mail:
    Free
  • Local DID:
    1 Free
  • Advanced Call-Forwarding:
    1 Free


signup now

SIP Trunking

SIP with Bundle 3000

£ 20.95 /Month

  • No Setup Fees
  • Free UK Calls
    3000 Minutes
  • Outgoing UK Calls
    0.02/minute
  • Increment:
    6 seconds
  • # Channles:
    One (1)
  • Voice Mail:
    Free
  • Local/Toll-Free DID:
    1 Free
  • Advanced Call-Forwarding:
    1 Free


signup now

SIP TRUNKS FROM AG TELECOM

At AG Telecom, we offer reliable and high-quality SIP trunks that enable you to make and receive calls over the Internet. Our SIP trunks are designed to work seamlessly with your existing PBX system, and we provide support for a wide range of SIP-compatible PBX systems. With our flat monthly rate, you can enjoy unlimited calling to anywhere in the world without having to worry about hidden fees or charges. Our SIP trunks are easy to set up and use, and we use secure connections to ensure the privacy and security of your calls. With AG Telecom, you can experience crystal clear voice quality, HD voice support, and exceptional customer service and support.

Sign Up today

For more information and a no-obligation quote please fill out the simple form below and one of our Customer Service Consultants will contact you and you can save upto 40% against these bundles

Frequently Asked Questions

Porting can vary depending on the losing carrier. Generally, porting takes about two weeks, but could potentially be longer. Providing all necessary porting documentation to your onboarding specialist will help speed this process along. You will require a signed LNP Form, as well as an Invoice from the losing carrier not more than 30 days old with all the numbers shown, including current billing name/address, and account number.

If you require rushed “HOT Porting” whereby ports are completed within 48 to 72 hours this an option available from the Porting Dept. However, there is additional costs associated with this which can be quoted and requires a signed LNP Hot Port Form.


Yes, we can do so by porting your existing numbers to AG Telecome. Simply put, porting means the transfer of numbers from one provider to another.  In this case, you would transfer your numbers over to AG Telecome. *not all numbers are portable

The cost for porting one (1) number is $10 per DID. Bulk porting will be quoted upon signup.


AG Telecome offers three SIP Trunk solutions. Direct SIP Trunk to IP-PBX, SIP Trunk to PRI, and SIP Trunk to Analog lines. So yes virtually every PBX is capable of using our SIP Services either directly, via a SIP to PRI gateway or a SIP to Analog gateway.

However, some PBX’s may require licensing to enable the feature. These costs are not covered or included in our delivery of SIP Services unless otherwise agreed to.


Session Initiation Protocol

Session Initiation Protocol (SIP) is a signaling protocol developed by the Internet Engineering Task Force (IETF). It can be used for VoIP as well as other types of multimedia communications. The details regarding the protocol are described in RFC 3261, although there are subsequent RFCs covering various functionality related to SIP including DTMF, Privacy, Early Media and more.

SIP Trunking is used to describe connectivity between a SIP enabled PBX or SIP enabled dialer and the carrier using Session Initiation Protocol (SIP). Most recent versions of popular PBX equipment including Avaya, Cisco and Nortel and predictive dialer equipment such as Aspect, Altitude, Asterisk, Vicidial and Interactive Intelligence, just to name a few, natively support SIP Trunking.

VerseTEL provides SIP Trunking services for Canada, USA, UK, Asia and internationally.

Simply put thought, SIP Trunking is using internet services to connect your Office Phone system to telephone Numbers (DID’s) be it cell phones, offices, or toll-free numbers.

It allows PSTN interconnection to anywhere in Canada – all from a single connection that supports a wide range of voice and data services.


VoIP – Voice over IP, is a set of technologies and protocols which allows transmission of voice over Internet Protocol (IP).This includes proprietary protocols such as Skype as well as open protocols such as SIP, IAX2 and H323.


“SIP Enabled” indicates whether the device or equipment such as A PBX or Dialer supports the Session Initiation Protocol (SIP). Most recent PBXs and dialers support SIP natively.

Star Telecom has successfully deployed our SIP Trunking, SIP DID, SIP Toll Free and SIP Long Distance services with the following:

  • Avaya
  • Cisco
  • Dialogic
  • Asterisk
  • Vicidial
  • Altitude
  • AudioCodes
  • FreePBX
  • 3CX