All-In-One Unified Communications
Audio, Video, Instant Messaging, WebRTC, Conferencing
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 the maximum of online extensions | 10 | Base on the license |
| Number of Simultaneous Calls Supported | 3 | Unlimited |
| Linux OS: CentOS, Debian, Ubuntu | Y | Y |
| Windows OS: Win 10, Windows Server 2008, 2012, 2016, 2018 | Y | Y |
| Docker container deployments | Y | Y |
| Cluster deployments | Y | Y |
| High Availability | Y | Y |
| Congestion Management | Y | Y |
| Multi-Tenant | Y | Y |
| Admin can switch to tenant role to manage the tenant's resource and switch back by one click, no need sign out and sign in again | Y | Y |
| Sending file, picture | Y | Y |
| Sending screenshot | Y | Y |
| Presence | Y | Y |
| HD audio call | Y | Y |
| HD video call | Y | Y |
| Rich text chat | Y | Y |
| Audio conference | Y | Y |
| Video conference | Y | Y |
| Conference recording | Y | Y |
| Dial in conference from trunk | Y | Y |
| Invite mobile phone or landline into conference | Y | Y |
| Transfer the call into conference | Y | Y |
| See all particiants in the conference | Y | Y |
| Text chat to participants in the conference | Y | Y |
| Sharing screen or application window to particpants in the conference | Y | Y |
| Conference management: Lock, mute, start recording, stop recording, kick participant, change video layout. | Y | Y |
| Sharing screen | Y | Y |
| Multi calls | Y | Y |
| Call Waiting | Y | Y |
| CDR | Y | Y |
| Custom global office hours | Y | Y |
| Custom extension office hours | Y | Y |
| Custom office hours for inbound rule | Y | Y |
| Custom office hours for outbound rules | Y | Y |
| Custom holidays | Y | Y |
| Exception route | Y | Y |
| Route call base on the callee status (busy, ready/not ready, offline, no answer within the time specified) | Y | Y |
| Route call base on the DID and CID | Y | Y |
| Disable inbound rule base on the custom office hours | Y | Y |
| Disable outbound rule base on the custom office hours | Y | Y |
| Route inbound call base on the specified office hours | Y | Y |
| Route extension call base on the specified office hours | Y | Y |
| Advanced route forward rules | Y | Y |
| Auto attendant | Y | Y |
| Voice Mail | Y | Y |
| Central phone book | Y | Y |
| Auto-Sync phone book to App | Y | Y |
| Call Hold / Call resume | Y | Y |
| PRACK | Y | Y |
| Early Media | Y | Y |
| Attended transfer | Y | Y |
| Blind transfer | Y | Y |
| MWI | Y | Y |
| Ring multiple device simultaneously | Y | Y |
| Pick up the ringing call | Y | Y |
| Pick up the held call | Y | Y |
| Support connect to the E1/T1 gateway, SIP Trunk, IMS SIP Trunk | Y | Y |
| Configure custom SMTP | Y | Y |
| Configure custom DNS Server | Y | Y |
| Configure custom Web domain | Y | Y |
| Configure custom SIP domian for each tenant | Y | Y |
| Configure custom transport ports | Y | Y |
| Multiple transports(WSS, TLS, TCP, UDP) | Y | Y |
| Busy Lamp Field (BLF) | Y | Y |
| Generate the call report | Y | Y |
| Audio call recording | Y | Y |
| Video call recording | Y | Y |
| Record the caller and callee voice in separated track(dual track recording) | Y | Y |
| Recording files management??list??query??download??play in web browser, delete?? | Y | Y |
| Intercom/ Paging | Y | Y |
| Confiture BLF | Y | Y |
| Web management UI | Y | Y |
| IP Phone Auto Provisioning | Y | Y |
| Configure the mobile app by scan QR code | Y | Y |
| Configure Windows desktop app by scan QR code | Y | Y |
| Configure the WebRTC client by scan QR code | Y | Y |
| Display status and statics on Web | Y | Y |
| Display all components status on Web | Y | Y |
| Log file management | Y | Y |
| Events viewer | Y | Y |
| Activity viewer | Y | Y |
| Integrated the Web Server | Y | Y |
| Backup and resotre by the snapshot of Virtual Machine or cloud server | Y | Y |
| VMware / Hyper-V | Y | Y |
| Public cloud(AWS, AZURE, GCE and so on) | Y | Y |
| Send voice mail by email | Y | Y |
| Send extension account configure information by email | Y | Y |
| Ring Group | Y | Y |
| Group/Call Queue Polling Strategy: Ring Simultaneously | Y | Y |
| Group/Call Queue Ring Simultaneously: Prioritized Hunt | Y | Y |
| Group/Call Queue Polling Strategy:: Cyclic Hunt | Y | Y |
| Group/Call Queue Polling Strategy:: Least Worked Hunt | Y | Y |
| Group/Call Queue Polling Strategy:: Paging/Intercom | Y | Y |
| Call queue / Call center / Contact center | Y | Y |
| Monitor the call | Y | Y |
| Whisper | Y | Y |
| Barge-in | Y | Y |
| Intercept | Y | Y |
| Exclusive Agent | Y | Y |
| VIP number (Will be answered 1st priority) | Y | Y |
| Two levels list of harassment, play alert voice to the caller who in the list | Y | Y |
| Lost calls query | Y | Y |
| Queue manager | Y | Y |
| Set multiple extensions a queue manager | Y | Y |
| Custom queue SLA time | Y | Y |
| Provide REST API to set the agent ready or not ready status | Y | Y |
| Allow / disallow set the agent status to not ready automatically after agent completed a call | Y | Y |
| Send alert email to the queue manager when a new lost call appears in the queue | Y | Y |
| Send alert email to the queue manager when the caller in the waiting queue reached SLA time | Y | Y |
| List the lost calls of the queue | Y | Y |
| Custom prompt file | Y | Y |
| Custom intro prompt file | Y | Y |
| Announce Queue position to waiting caller | Y | Y |
| Custom maximum callers waiting in the queue | Y | Y |
| Custom the interval for Announce Queue position | Y | Y |
| Custom the queue SLA time | Y | Y |
| Search call recording | Y | Y |
| Android App | Y | Y |
| iOS App | Y | Y |
| Windows Desktop App | Y | Y |
| WebRTC Client App | Y | Y |
| macOS App | Y | Y |
| Android SDK | Y | Y |
| iOS SDK | Y | Y |
| Windows SDK | Y | Y |
| WebRTC Client SDK | Y | Y |
| macOS SDK | Y | Y |
| Mobile client app push notificaitons | Y | Y |
| Avoid the NAT issues automatically | Y | Y |
| Auto provision IP phone by PnP, RPS, SBC | Y | Y |
| Reboot Phone remotely | Y | Y |
| Support popular IP Phones(HTEK, FANVIL, YEALINK, GRANDSTREAM, SNOM) | Y | Y |
| Disallow the internal call by country code | Y | Y |
| Process E164 number | Y | Y |
| Add the IP into blacklist automatically base on the failed authoriation times | Y | Y |
| Add the IP into blacklist automatically base on the IP packets rate | Y | Y |
| Allow add the IP into IP blacklist and whitelist manually | Y | Y |
| Music On Hold | Y | Y |
| Set the music files list for Music On Hold | Y | Y |
| Play the music files randomly for Music On Hold | Y | Y |
| Add / manage the blocked numbers | Y | Y |
| Action URL for the Auto attendant | Y | Y |
| Route the call base on the Action URL result | Y | Y |
| Billing base on the called number prefix | Y | Y |
| Limited call to national and internation if no enogh balance | Y | Y |
| Billing to outbound call | Y | Y |
| Billing to inbound call | Y | Y |
| Custom billing rule: Connect fee, Grace Period, Free Seconds, Interval 1, Price 1, nterval N, Price N, Post Call Surcharge base on percent) | Y | Y |
| REST API | Y | Y |
| WebSocket Publisher | Y | Y |
| Publish the extensions events to WebSocket Subscriber(register, unregister, start call, call ringring, call answered, call hangup, call failed, call hold/unhold) | Y | Y |
| Publish CDR to WebSocket Subscriber once a call is completed | Y | Y |
| Publish the trunk status to WebSocket Subscriber (Trunk connected, trunk disconnected) | Y | Y |
| Publish the queue status to WebSocket Subscriber(the call push into the queue, the call lost from the queue, the call is pop up from the queue, the call is answered by agent, the call is failed) | Y | Y |
| Web Hook(Send the extension events, CDR events to the Web URL) | Y | Y |
| Outbound caller ID | Y | Y |
| Set special Outbound caller ID for emgency call | Y | Y |
| Set Outbound caller ID for extension | Y | Y |
| Set Outbound caller ID for conference | Y | Y |
| Set Outbound caller ID for call queue | Y | Y |
| Enable/disable display extension password in browser | Y | Y |
| Extension group | Y | Y |
| Extension profile | Y | Y |
| Save extension QR code from web page | Y | Y |
| Update Extension balance | Y | Y |
| Extension group management | Y | Y |
| Multiple extension group management | Y | Y |
| Update extension permissions by manage the extension group | Y | Y |
| Allow/disallow the extensions which in the specified extension group make the outbound call | Y | Y |
| Allow/disallow the extensions which in the specified extension group access web management | Y | Y |
| Extension can sign in the Web Management by enter extension number or email | Y | Y |
| Allow/disallow extension delete his recording files | Y | Y |
| Allow/disallow tenant/extension sign in the Web Management cocurrent from multiple devices | Y | Y |
| Allow custom maximum times of failed authorization for sign in Web Management | Y | Y |
| Allow custom maximum time duration of IP blocked if the IP was blocked due to failed authorizatio times exceeds when sign in the Web Management. | Y | Y |
Top Reasons For Choosing PortSIP
Not forked from open source PBX
PortSIP PBX is developed by PortSIP team and isn’t forked from any open source VoIP PBX, which means much better performance and features.
Video conferencing
PortSIP PBX has built-in MCU, is an unparalleled, SIP standards video conferencing and collaboration solution that transforms your communications, increases productivity, slashes travel expenses, boosts customer service and propels sales!
Professional and realtime support
PortSIP has ability to provides the professional and realtime support since the PBX system is developed by own team, no dependencies on other third party.
Client SDK and Mobile App
PortSIP also provides the client VoIP SDK and Mobile App, offers customization and rebranding service with one times PortSIP PBX license fee.
Customizable
PortSIP provide customization service that allows users to add new features according to their requirements, user could has unique PBX as to facilitate their business.
WebRTC Gateway
PortSIP PBX integrated the WebRTC Gateay which allows you make & receive calls or join video conference from web browser, without the need to download any plugins, bridge the communications between Internet and SIP.