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Cloud Telephone Solutions

  • Adaptable to all business environments
  • Flexibility in usage, resizing, and accessibility
  • No size limitation
  • No busy signal
  • No major capital investment is required to use the service
  • No maintenance cost
  • Kari’s Law and Ray Baum’s Act Compliant
  • Increased features beyond what typically is available in expensive systems
  • High quality, reliability, and better than Military grade security
  • Dynamic encryption between server and phone (before and during a call)  
  • Microsoft Teams® Integration (don’t need a Teams enabled IP phone) 
  • Unlimited North American Calling and Low International Calling rates

Why Converged Networks is better?

Click the button to the right to see why Converged Networks Cloud Telephony Solutions are better then anyone else and how we compare with them and a physical PBX.

Click one of the icons below to request a quote; to see a brochure; view a presentation or look at the phones we offer

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Request A Quote

Request A Quote

Brochure

Brochure

Presentation

Presentation

SIP Phones

SIP Phones

Features of our Cloud Telephony Solutions

Auto Attendant
  • 2, 3, 4 and 5-digit extension dialing
  • Name playback after extension dialing 
  • Blocking of certain numbers 
  • Call redirection based on user input 
  • Call redirection on timeout 
  • Call redirection on time-of-day (day/night mode) 
  • Call redirection on loss of registration 
  • Call disconnect on no activity 
  • Multiple language support (e.g. French/English) 
  • Dial by name (directory) 
  • Pre-recorded standard greetings (e.g. for support, for accounting, …) 
  • Greeting recording from extension 
  • Upload and playback of greetings through web interface 
Conferencing
  • Scheduled and spontaneous conferences
  • Multiple language support 
  • User/moderator mode 
  • Email invitation, participation email (calendar format) 
  • Voice activity detection in conference 
  • Multiple, independent conference rooms 
  • Name recording and conference announcement 
  • Conference recording and download through web interface 
  • Attended and blind transfer into conference 
Cell Phone
  • Parallel forking to cell phone (single/multiple cell phones or external PSTN phones), immediate and after delay
  • Parallel forking to cell phone in group calls (queues, hunt groups)
  • Direct inward station access, outbound calls from cell phone
  • Call back for outbound calls
  • Call back on mailbox message
  • Call back when extension becomes available
Queues
  • Agent login/logout (static/dynamic), automatic logout 
  • Escalation agents 
  • Minimum staffing for logout prevention 
  • Ring melody indication 
  • Queue priority for agent arbitration 
  • Call volume throttling (maximum number of parallel active calls) 
  • Programmable caller-ID manipulation 
  • Address book integration 
  • Caller-ID based agent preference routing 
  • Programmable event for connecting the call 
  • Programmable ring back tone 
  • Selectable music on hold 
  • Programmable agent idle time 
  • Queue position announcement 
  • Daily activity summary email reports 
  • Call recording of queue calls 
  • Multiple agent selection algorithms 
  • Multiple call redirection events and destinations 
  • User input key handling 
  • Dynamic call routing (e.g. day/night mode) 
  • Web-based agent activity tracking 
  • Web-based call monitoring 
  • Customizable email on connect to the connected agent 
  • Call whisper mode, call listen in, call barge in 
  • SOAP queue information and control
Hunt Groups
  • Three stage hunt group with parallel and sequential forking
  • Call routing with night/day mode and other flags
  • Call routing to final stage destination (internal or external)
  • Customizable ring melody
  • Daily CDR reporting
  • Programmable music on hold source
  • Call recording of hunt group calls
Mailbox
  • Multiple prompts and name recording (from extension or web interface)
  • Voicemail to email
  • Moving and copying of messages to extensions and groups
  • Recording of comments to messages
  • Fast forward and rewind for message playback
  • Envelope information
  • Callback from mailbox
  • Message waiting support (MWI)
  • Call recording into mailbox
  • PIN code-based log in
  • Star code access from mailbox
  • Mailbox sharing
  • Voicemail to email with or without attachments
Extension Features
  • Extension specific dial plan, time zone and language support
  • Multiple extension alias names, ANI assignments
  • Hot desking
  • Park orbit and pickup preference
  • Call redirection on busy, not registered, on timeout, always
  • Daily email support
  • Picture upload
  • Camp on, call waiting enable/disable, caller-ID blocking
  • Emergency dialing support, EPID support for E911
  • Emergency call recording
  • Intercom (2-way) support
  • Call number limitation
  • Multiple registrations per extension
  • Call screening (anonymous, all) with name recording
  • Email (missed calls, CDR, voicemail, status changes, mailbox full)
  • Personal address book
  • Plug and play, based on MAC and/or username/password
  • Extension button programming
  • Prepaid balance
  • Shared Line emulation
  • Extension, queue, hunt group and status flag monitoring
  • DND, agent status
IVR
  • Regular expression-based routing for DTMF input and Caller-ID
  • Timeout handling
  • SOAP server-based routing
  • Uploading of WAV files
  • Address Book
  • User address book and domain address book
  • Black and white list support
  • CSV Import
  • From/To-header rewrite on incoming calls
  • Speed dial support
Paging
  • SIP, unicast and multicast paging
  • Connection confirmation
  • Plug and play integration
  • Recorded and immediate paging
Security Features
  • SIP TLS transport layer
  • Domain specific certificates
  • SRTP media encryption
  • HTTPS provisioning
  • Selectable password policy
  • Automatic blacklisting after unsuccessful authentication attempts
  • Explicit black and white listing of IP addresses
Networking Support
  • Near-end and far-end NAT traversal with mini session border controller
  • Multiple IP-address routing in SIP (e.g. private and public address mix, VPN)
  • IPv4 and IPv6 support
  • Multiple SIP ports, multiple HTTP ports, multiple TFTP ports
  • QoS definition
Web Interface
  • Sample text. Click to select the text box. Click again or double click to start editing theAccess from anywhere
  • User, domain and system mode login
  • Click-to-dial
  • Secure transport, provisioning of certificates
  • text.
Provisioning
  • Template-based provisioning of popular
SIP devices
  • Provisioning of firmware
  • Multicast plug and play
  • Reboot from web interface
  • Provisioning of XML-page access for address book
SIP-Trunking 
  • Registration and Outbound Proxy modes for ITSP and PSTN gateways
  • Caller-ID indication (RFC 3325-based and Remote-Party-ID)
  • Inbound call routing
  • CO-line emulation
  • Microsoft Exchange Support
Dial Plans
  • Multiple dial plans per domain
  • Failover support
  • Dial plan-based routing to extensions
  • PIN code authentication for outbound calls
  • Client matter code enforcement for outbound calls
Administration
  • Email Alerts on various events
  • Multiple domains
  • Backup/restore function
  • Active CPU load monitoring
CRM Integration
  • CSTA support
  • Integrated TAPI Service Provider (TSP)
  • Usage of favorite address book tools
Media support 
  • Inband and out-of-band DTMF detection
  • Narrowband G.711, GSM full-rate and G.726 codec support
  • Support for G.729A (requires additional License)
  • Support of wideband (G.722)
  • Transparent codec pass-through
  • T.38 pass-through