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The C600 provides the complete desk phone solutions with
flexible functionality, HD Voice Quality, and highly easy
operation. It is absolutely an excellent Smart Video Phone for
small to large sized business, as well as mission-critical
Enterprise Business.
Highlights
√ Full Programmable DSS Key and Soft Function Key
√ DSS Can Support Upto 100 Stations
√ On Line Recording & Message Forwarding
√ Call Conference Enhancement
√ Flexible Dial Plan with Auto Provision
√ Support LDAP & XML Phone Book
√ Support WIFI / Bluetooth (Optional)
Display 7” TFT 800x480 Capacitive Multi Touch ScreenCall Features Call Forwarding Call Transfer (blind/attended/alert) Call Holding Call Waiting Call Conference BLF List Capable of 10 way conversation Join Call Pickup Call Completion Hot desk function Soft DSS Key (Upto 100) Auto Redial / un-redial Support multi line and pre-dial Support messaging and MWI Voice recording during talking /local Flexible dial plan Hotline/Warm-line Call Rejection Black List Barring function for outgoing calls Do Not Disturb Auto Answer (Hands- free / Headset) Caller ID display CLIR (rejects anonymous calls) CLIP (to make an anonymous call) Dial without registration Supports Call Logs with Missed / Incoming / Outgoing calls Intercom/Intercom barge Password dial Headset ring Direct IP call without SIP proxyPhone features Android 4.2 OS Supports high quality video call Supports SIP 2.0 (RFC3261) and correlative RFCs Supports 6 SIP servers and Backup SIP proxy servers. Supports SIP UDP/TCP/TLS Supports HDMI Supports USB Host Supports Acoustic echo cancellation (AEC) - 128ms max filter length in duplex speaker phone mode Supports PLC & adaptive jitter buffer Audio Features Wideband codec: G.722 Narrowband codec: G.711(A/μ), G.723.1, G.729AB, iLBC, AMR DTMF: In-band, Out-of-band(RFC 2833) and SIP INFO Full-duplex hands-free speaker phone with AEC (Max filter length – 128ms) Voice Activity Detection (VAD) Comfort Noise Generation (CNG) Adaptive jitter buffer Packet loss concealment (PLC) Audio format: WAV/MP3/OGG |
Video Features Video codec: H.264 / H.263 Image codec: JPEG/PNG/BMP/GIF Video format: MP4/3GP/FLV Video call resolution: QCIF / CIF / VGA / 4CIF (1280x720P Optional) Bandwidth selection: 64kbps~4Mbps Frame rate selection: 10~30fps Picture-in-Picture (PIP) Video from remote site can be displayed in full screen Supports up to 4 video display mode Network/ Security Features WAN/LAN: support bridge mode Supports PoE (802.3af) Supports main DNS and secondary DNS server Supports VLAN Supports SNTP Client Supports VPN L2TP / PPTP / IPSec Supports SIP SRTP, Supports Web HTTP / HTTPS Supports QoS: 802.1p/q, DSCP Supports MD5 authentication Supports Web Filter DHCP/ static/ PPPoE STUNMaintenance & Management Android 4.2 Supports third party communication APP Supports Web, Telnet Web Management with different account right Supports automatic upgrades/ configuration deployment Supports encrypted configuration files download with AES Supports pushing message Supports Auto-Provisioning (DHCP option/ PnP/ Phone flash) Supports TR-069 Supports Web upgrade Supports U Disk/SD upgrade Supports backup/restore/factory reset dataPhysical Features Adapter Input:100-240V Adapter Output:12V/1A WAN Port -10/100/1000 Base-T RJ-45 for LAN LAN Port- 10/100/1000 Base-T RJ-45 for PC HDMI Port - Type A SD Interface - TF Card Support Upto 32G USB Port- USB 2.0 Power Consumption- Idle: 2.5W/Active: 5W LCD Size Diagonal:7 inch (800 x 480) Capacitive touch screen Camera- Adjustable Position Operation Temperature: 0~40℃ Relative Humidity: 10～65% CPU Freescale Core Quad 1 Ghz SDRAM - 1GB DDR3 1066 Flash- 4GB Weight- Phone:1.2Kg/Total:1.8Kg |
| Call Forwarding Call Transfer(blind/attended/alert) Call Holding Call Waiting Call Conference BLF List Capable of 10 way conversation Join Call Pickup Call Completion Hot desk function Soft DSS Key (Upto 100) Auto Redial / un-redial Support multi line and pre-dial Support messaging and MWI Voice recording during talking /local Flexible dial plan |
Hotline/Warm-line Call Rejection Black List Barring function for outgoing calls Do Not Disturb Auto Answer (Hands- free / Headset) Caller ID display CLIR (rejects anonymous calls) CLIP (to make an anonymous call) Dial without registration Supports Call Logs with Missed / Incoming / Outgoing calls Intercom/Intercom barge Password dial Headset ring Direct IP call without SIP proxy |
System Requirements
A Windows desktop or Windows server OS: Windows 7/8, Windows 10, Windows Server 2008 R2 with SP1, 2012 R2, 2016 R2) IPv4/IPv6 Windows firewall
This document assumes that the Windows OS is already deployed and administrators of PortSIP PBX are allocated the administrator permission to Windows.
Hardware and Software Dependencies
OS Supported by PortSIP PBX includes:
Linux Server:
CentOS 7 or higher, 64bit; gcc/g++ 6.4 or higher
Ubuntu 16.04.4 or higher, 64bit; gcc/g++ 6.4 or higher
Debian 9.0 or higher, 64bit; gcc/g++ 6.4 or higher
Windows Desktop:
Windows 7, 8 and 10, 64-bit
Windows Server:
Windows 2008 R2 SP1, 2012 R2, 2016 R2, 64-bit
Important: The OS must be up to date.
Cloud and Virtualization Environment Supported
To build high-availability communication solution to help clients reduce cost and improve communication performance, PortSIP PBX commits support on cloud services and have confirmed availability on following cloud and virtualization environment:
- VMware ESX 5.X and above.
- Linux HyperV
- Microsoft HyperV 2008 R2 and above
- Amazon AWS
- UCloud
- Alibaba Cloud
- Linode
- Digital Ocean
- Godaddy VPS and Cloud
- Tencent Cloud
System performance depends on following key factors:
- Maximum simultaneous calls needed for PBX
- Maximum online users needed for PBX
- Recordings for calls
- Record audio only or both of audio, video
- Maximum online users for audio/video conferences on PBX
- Maximum IVR (Virtual Receptionist) on PBX
- Maximum Call Queues on PBX
- Maximum Ring Groups on PBX
Depending on the key features listed above, PortSIP PBX is able to run on PCs and servers with various CPSs ranging from Intel i3 CPU to Inter Xeon
Other Requirements
- Latest Firefox, Google Chrome or Internet Explorer
- Microsoft .NET Framework version 4.5 or higher
- Knowledge of Linux and Linux Internet administration
- Knowledge of Windows and Windows Internet administration
- A constant internet connection to service.portsip.com on port 6881.
- A constant internet connection to stun.portsip.com and stun1.portsip.com on port 3478.
- A constant internet connection to stun4.l.google.com on port 3478.
PortSIP PBX Features
PortSIP PBX free Edition and Full Edition are equipped with same features, with the only difference that the free Edition only support up to 3 simultaneous calls only.
| Features | Free Edition | Full Edition |
| Extensions | Unlimited | Unlimited |
| Number of Simultaneous Calls Supported | 3 | >10,000 |
| Linux Support (CentOS, RHEL, Debian, Ubuntu) | Y | Y |
| Multi-Tenant | Y | Y |
| Call Logging | Y | Y |
| Call Forward on Busy or No Answer | Y | Y |
| Call Routing by DID | Y | Y |
| Auto Attendant / Digital Receptionist | Y | Y |
| Voicemail/ Music on Hold | Y | Y |
| Central Phonebook | TBD | TBD |
| Call Transfer | Y | Y |
| MWI – Message Waiting Indicator | Y | Y |
| Ring Extension & Mobile Simultaneously | Y | Y |
| Automatic Pickup on Busy | Y | Y |
| Supports SIP Trunks/ Gateways | Y | Y |
| Custom SMTP Server | Y | Y |
| Custom FQDN | Y | Y |
| Busy Lamp Field (BLF) | Y | Y |
| Call Reporting | Y | Y |
| Call Parking / Pickup | Y | Y |
| Call Queuing | Y | Y |
| Audio Call Recording | Y | Y |
| Video Call Recording | Y | Y |
| Intercom/ Paging | Y | Y |
| Call Recordings Management | Y | Y |
| Configure BLF’s from the Clients | Y | Y |
| Web-based Management Console | Y | Y |
| Automated Provisioning of Devices | Y | Y |
| Real Time Web-based System Status | Y | Y |
| Integrated Web Server | Y | Y |
| Easy Backup and Restore | Y | Y |
| VMware / Hyper-V Compatibility | Y | Y |
| Scheduled Backup | Y | Y |
| Scheduled Restore | Y | Y |
| Inbuilt Fail Over Functionality | Y | Y |
| Standby License | Y | Y |
| Deploy as cluster | Y | Y |
| See the Presence of Your Colleagues | Y | Y |
| Receive Voice Mail via Email | Y | Y |
| Advanced Forwarding Rules | Y | Y |
| Setting Up Conference Calls | Y | Y |
| Click2Call | Y | Y |
| View Presence of Remote Offices | Y | Y |
| Advanced Queue Strategies | Y | Y |
| Advanced Call Reporting | Y | Y |
| Real Time Queue Statistics | Y | Y |
| Ability to Use PortSIP VoIP SDK | Y | Y |
| Real Time Queue Monitoring | Y | Y |
| Call Recordings Search | Y | Y |
| Supports External Agents | Y | Y |
| Android Client (Provide rebrand/OEM) | Y | Y |
| iOS Client (Provide rebrand/OEM) | Y | Y |
| Windows Client | Y | Y |
| Mac Client | Y | Y |
| Web client | Y | Y |
| Mobile PUSH | Y | Y |
| Avoid NAT Problems | Y | Y |
| Automatic Plug & Play Phone Provisioning | Y | Y |
| Manage IP Phones Network Wide from Console | Y | Y |
| Restart Phones Remotely | Y | Y |
| Supports Popular SIP Phones | Y | Y |
| Provide Client VoIP SDK(Android, iOS, Windows, macOS) | Y | Y |
| Plugin Free – WebRTC | Y | Y |
| One-click conference | Y | Y |
| Meeting Recording | Y | Y |
| Unlimited Users | Y | Y |
| Meeting Participants Included | 3 | 100 |
| Full REST API | Y | Y |
| Security | Y | Y |
| E164 Number processing | Y | Y |
Agenda