SipX VOIP Phone PBX Services and Consultants
Drouillard & Associates offers business-grade IP PBX built on open source software not built for geeks but enterprise users requiring easy to use, stable and scalable solutions for mission critical applications. Our system can meet the communications needs ranging from 10 people offices with 4 analog trunk lines to companies with thousands of users. Open source offers lowest total cost of ownership (TCO) and it does not lock you into a vendor specific and proprietary solution that grows with your business.
Deploying VOIP is easy and installation is usually done in hours. Because of its Web based administration interface as well as the ability of VOIP to put the users in control, ongoing system administration is easily done by a receptionist or similar person. The following examples are intended to give some typical and therefore recommended system configurations. They range from very small deployments at small offices with a few users up to very large and therefore fully redundant deployments serving several thousand users.

VOIP provides a complete solution to your enterprise telephony needs. The system's architecture lets you easily distribute servers, gateways and intelligence strategically on your network, within one office or among branch offices, for cost savings, high reliability, backup and load balancing. Capabilities such as automatic trunk fail-over and redundancy, high-availability, and least cost routing are must-have features in today's environment.

For large companies, VoIP also offers some very unique possibilities. Many larger companies have already switched to VoIP or have plans to do so. The cost of a VoIP system compared to a standard PBX system is a fraction of the cost. Whereas a standard PBX systems starts well above $5000, that same VoIP installation, with even more features, can be done in many cases for under $1000. Some companies utilizing the technology by conducting all intra-office calls through a VoIP network. Because if the network is wired properly (including possibly making use of fiber optic connectivity) the quality of sound far surpasses that of analog service. Some international companies are using VoIP to circumvent the high cost of international calling. Included in this is the ability for the company or it's customers to call a local number and have it routed through VoIP to the country where they also have an office and then hoping the call off of the internet onto the public network in that local country. This allows the company and their customers to pay local rates internationally. It also allow companies with multiple office to utilize the VoIP network and have all inter-office calls, no matter what office each person is in to as if they are calling someone in the next cubicle.
While your current long-distance plan covers you for only one location. With VoIP no matter what type of a device you use you can make a call from anywhere where you can get a broadband connection. That is because all three methods mentioned above, unlike analog calls, send the call information via the Internet. Hence, you can make calls from home, on vacation, on business trips and almost anywhere else. With VoIP, you can bring your home phone along with you anywhere you go. In the same way, computer-to-computer connections imply that as long as you have your laptop and a broadband connection, you are ready to go.
sipXecs Telephony System Features
- Transfer (consultative & blind)
- Call coverage
- Call hold / retrieve
- Consultation hold
- Music on Hold for IETF standards compliant phones
- Uploadable music file
- 3-way conference
- Call pickup (global and directed call pickup)
- Call park & retrieve
- SIP URI dialing
- CLID (Calling Line Identification)
- CNIP (Calling party Name Identification Presentation)
- CLIP (Call Line Identification Presentation)
- CLIR (Call Line Identification Restriction)
- Per gateway CLIP manipulation
- Call waiting / retrieve
- Do not Disturb (DnD)
- Forward on busy, no answer, do not disturb
- Multiple line appearances
- Multiple calls per line
- Multiple station appearance
- Outbound call blocking
- Click-to-dial (Windows)
- Redial
- Call history (dialed, missed, received)
- Auto off-hook / ring down
- Incoming only
- Configuration of individual Speed Dial softkeys
- Auto-generation of Directory information
User Self-Control
- Every user on the system gets access to a personal Web user portal for self-management and control
- Management of voicemail
- Configuration of unified messaging preferences
- Time based find-me / follow-me
- Flexible configuration of call forwarding
- Personal call history
- Personal phone book, speed dial and presence management
- ACD presence and supervision capabilities
- Individual phone management
Superior Voice Quality
- Peer-to-peer media routing with quality optimization
- Lower delay and jitter
- Support for any codec supported by the phone
- Codec negotiation directly between phones
Dial Plan
- Easy to use GUI based dial plan manipulation
- Time-based dialing rules with different admin defined schedules
- Rules based least cost routing
- Automatic gateway redundancy and failover
- Specific E911 routing
- Permission based rules
- Prefix manipulation
- Dial plan templating for international dial plans
User Management
- Create a user, provision a phone and assign a line in only three clicks – easy!
- Numeric or alpha-numeric User ID
- User PIN management (UI or TUI)
- Aliasing facility (numeric and alpha-numeric aliases)
- Extension and alias uniqueness assurance
- Granular per user permissions
- Call permissions:
- 900 Dialing
- International Dialing
- Long Distance Dialing
- Mobile Dialing
- Local Dialing
- Toll Free Dialing
- Forward Calls External
- System permissions:
- User has voicemail inbox
- User listed in auto-attendant directory
- User can record system prompts
- User has superuser access
- User allowed to change PIN from TUI
- Custom permissions added by the administrator
- Supervisor permission for groups (e.g. Call Center supervisor)
- SIP password management for security
- User groups with group properties
- Per user call forwarding (find me / follow me)
- To local extension, PSTN number, or SIP address
- Scheduled forwarding based on user defined individual schedules
- Parallel or serial ring
- Allows definition of ring time before trying next number
- Allows several forwarding destinations
- Follow-me configuration using user portal
- Extension pool with automatic assignment
- Per user Caller ID (CLID) assignment
- Per user Caller ID blocking
PSTN Trunking
- Unlimited number of PSTN gateways and trunk lines
- Direct Inward Dialing (DID)
- Local DID per gateway
- DNIS
- CLIP Management
- User CLIP
- Gateway default CLIP
- Prefix stripping / appending
- Per gateway CLIR
- Automatic Route Selection (ARS)
- Least-cost routing (LCR)
- Automatic failover if unavailable
- Automatic failover if busy
- FAX support (pass-through)
- Mixing of PSTN trunks with SIP trunks
Performance
- Unlimited number of simultaneous calls
- Unlimited number of trunk lines
- 54,000 BHCC, 120,000 BHCC redundant(dependant on server platform)
- Up to 10,000 users per dual-server HA system
- Automatic time distribution of re-registrations
High Availability
- Optionally fully redundant call control system
- Based on DNS SRV (no cluster required)
- Load balance under normal operating conditions
- Geographic dispersion of redundant systems
- Real-time synchronization of state information
- Reports on load distribution
Security
- All outbound calls authenticated
- DoS attack prevention
- HTTPS secure Web access
- Secure user SIP password management
- TLS based signaling for SIP trunks (requires session border controller)
System Administration Features
- Browser based configuration and management
- LDAP integration (OpenLDAP)
- Integration with Microsoft Exchange 2007 for voicemail and Active Directory
- SOAP Web Services interface
- CSV import and export of user and device data
- Integrated backup & restore
- Scheduled backups
- Diagnostics
- Display active registrations
- Display job status
- Status of services
- Snapshot logs for debugging
- Logging (customizable log levels, message log per service)
- Domain Aliasing
- Support for DNS SRV
- Automatic restart after power failure
- Server statistics (integrated graphs and SNMP)
- Login history report (successful and unsuccessful)
Plug & Play Device Management
- Plug & play management of phones (see the list of plug & play managed devices)
- Plug & play management of PSTN gateways (see the list of plug & play managed devices)
- Auto-generation of phone / gateway config profiles
- Auto-pickup of profile by the phones / gateways
- Centralized management of all the parameters
- Centralized backup and restore of all configurations
- Auto-generation of lines by assigning users to devices
- Device group management & properties
- Firmware upgrade management
- Auto-discovery of phones and gateways
Voicemail Subsystem
- Integrated voicemail system at no extra cost
- Browser based user portal for voicemail management
- Message Waiting Indication (MWI)
- User configurable distribution lists
- Manage Notifications:
- Email notification of new voicemail messages
- Forwarding of message as .wav file
- Manage folders: Folders for message organization
- Manage greetings: Multiple customizable greetings
- Operator escape from anywhere
- Remote voicemail access
- Unlimited number of inboxes
- Up to 60 virtual media server ports per server
- Message store only limited by disk size
- Auto-removal of deleted messages
- Daily report on disk usage sent to admin
Personal Auto Attendant
- User configurable personal auto-attendant for every user on the system
- Individual zero-out to a personal assistant or receptionist
- Individual selection of language
- Personal greeting
Auto Attendants
- Unlimited number of auto-attendants
- Customizable IVR menus
- Dial by extension and name
- Night and holiday service
- Special auto-attendant
- Transfer on invalid response
- Nested auto-attendants (multi-level)
- Fully customizable actions: Operator, Dial by Name, Repeat Prompt, Voicemail login, Disconnect, Auto-Attendant, Goto Extension, Deposit Voicemail
- Uploadable custom prompts
- Configurable DTMF handling
Hunt Groups
- Unlimited number of hunt groups
- Serial and parallel forking (rings sequentially or at the same time)
- Configurable ring time per attempt
- Enable / disable user call forwarding rules while hunting
- Flexible configuration of destinations if no answer
Call Park Server
- Unlimited number of park orbits
- Visual indication on the phone of the state of the park orbit using the presence server
- Music on park
- Configurable call retrieve code
- Configurable call retrieve timeout
- Automatic park timeout
- Configurable park escape key
- Allow multiple calls on one orbit
Call Center Server (ACD)
- Supports several ACD servers, optional on separate server hardware
- Several (unlimited) queues per server
- Several lines per queue
- Support trunk lines (many calls per line) or single call per line
- Dedicated overflow queues or overflow to hunt group, extension or voicemail
- Configurable call routing scheme per queue:
- Ring All
- Circular
- Linear
- Longest idle
- Agent barge in (early termination of welcome message if agent becomes available)
- Agent presence monitor using presence server
- Separate welcome and queue audio
- Call termination tone or audio
- Configurable answer mode
- Agent wrap-up time configurable per queue
- Auto sign-out of agents if calls are not answered
- Configurable maximum ring delay
- Configurable maximum queue length
- Configurable maximum wait time until overflow condition
- Unlimited number of agents per queue
- Real-time Statistics:
- Agent statistics
- Call statistics
- Queue statistics
- Supervisor authorization for agent monitoring per group
- ACD historic reports for agents, calls, queues
- All reporting stored in database for post-processing if needed
Group Paging
- Integrated group paging server
- Unlimited number of paging groups
- Supports regular SIP phones using auto-answer
- Supports dedicated in-ceiling devices (SIP)