3CX CALL CENTER
Slash Your Phone
Bill by 80%
Save up to 80% on your telco bill. Get 3CX Free Through AG Telecom!With 3CX Call Center, you can reduce your phone bill by up to 80% with the use of SIP Trunking and VoIP technology. This means you can save more money on communication costs while enjoying high-quality voice and video calls.
- Support SIP Trunks
- Free Video Conferencing
- iOS and Android Apps
- Free Video Calling
Unified
Communications
Unified Communications Presence, Chat, Voicemail, Fax 2 Email
- Real-time Presence
- Voicemail to Text & Email
- Web Chat
- Fax 2 Email
Web Conference
With 3CX Call Center, you can easily conduct web conferences with your team members, customers, or suppliers. This feature allows you to collaborate in real-time, share files, and communicate effectively from anywhere in the world.
- Eliminate expensive web Conferencing Services
- No Monthly subscription fees
- Save on call conferencing costs
- No Expensive Cameras to buy
Office Without
Limits
3CX Call Center provides you with the flexibility to work from anywhere, whether you are in the office or on the go. With its mobile app and web client, you can easily stay connected to your phone system, colleagues, and customers from anywhere with an internet connection.
- Android Phones
- iOS Phone
- Android Tablets
- iOS Tablets
Why Choose 3CX With AG Telecom
Web Console
– Manage your call center from a user-friendly web interface.
View Contacts
Access all your contacts in one place for quick and easy reference.
Dashboard
Monitor your call center performance in real-time with an easy-to-use dashboard.
Agent Status Display
Know the status of your agents at a glance, including who is on a call, available or away.
Call Details
Get detailed information on every call for better tracking and analysis.
Instant Escalation
Quickly escalate calls to supervisors or managers when needed.
Call Back
Allow customers to request a call back instead of waiting on hold.
Call Recording
Record calls for training, quality control, and compliance purposes.
Customer Service with Contact / Call Center Software
3CX Call Center includes powerful contact center software that enables you to provide excellent customer service to your customers. You can easily manage incoming calls, route them to the right agent, and provide personalized service.
Never Miss a Call – Advanced Contact Center Reporting
Advanced Contact Center Reporting: With 3CX Call Center, you can track and monitor all incoming calls to ensure that no customer call is missed. Its advanced reporting feature allows you to analyze call patterns, agent performance, and customer feedback to improve your overall customer service.
- Integrated Wallboard for real-time monitoring
- SLA and Callback Statistics
- Detailed reports of longest wait time and abandoned calls
- Call Back option for customers not willing to wait
Advanced Queue Strategies and Real-Time Statistics
3CX Call Center offers advanced queue strategies that enable you to prioritize calls based on your business needs. You can also view real-time statistics that provide you with insights into agent activity, queue status, and customer satisfaction.
- Log agents in and out of queues
- Call Back – callers can hang up and keep their position
- Hunt by Threes – Random & Prioritized
- Round Robin
Superior Call Center Features Included
With 3CX Call Center, you get a complete suite of call center features such as call recording, call queuing, IVR, and more. These features enable you to provide high-quality customer service and improve your overall business communication.
- Listen in allows you to listen to a call without the caller or agent knowing
- Agents making a mess of a call? Use Barge in to take over
- Train new agents during a live call with the whisper function
- Live chat & talk with your website visitors
Server-side CRM Integration
3CX Call Center integrates with popular CRM software such as Salesforce, Microsoft Dynamics, and more. This integration allows you to streamline your business communication and provide personalized service to your customers.
- Zendesk, Salesforce, Freshdesk, Office 365, Microsoft Dynamics and more
- Launch phone calls with a single click from your browser or application
- Automatically link incoming calls to customer records
- Ensure customer satisfaction with quick and easy logging of interactions
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.
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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.
Can I Keep My Current Number(S)?
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.
Can I Use Your Service With My Existing PBX?
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