This Frequently Asked Questions (FAQ) document provides concise answers and guidance for using the UCM6300 Series IP PBX. It is intended to assist administrators and users in resolving common issues and understanding key features.
Product Overview & Specifications
The basic telephony software is based on Asterisk 16. However, a number of Grandstream’s proprietary voice/video algorithms, advanced video collaborations, device provisioning and remote management features are also integrated into the UCM6300 series.
Yes. Please see examples to illustrate the relationship below.
By default, UCM6300 series has “Opus Fullband (48kHz)” enabled, which will use Opus 48kHz sampling rate. If the users prefer more than 4 video feeds or more attendees in the meeting, please log in UCM6300 series web UI as admin, navigate to Basic Call Features→Multimedia Meeting→Meeting Settings and configure the “Meeting Audio Quality” option to “Opus Narrowband (8kHz)” instead.
Model | Video feeds/screen sharing | 1080p Video Quality (1 feed only) | Voice codec | Max attendees |
UCM6301 | 4 video feeds + 1 screen sharing | Yes | Opus (48KHz sampling rate) | 45 |
Opus (8KHz sampling rate) | 48 | |||
G.722 (16KHz sampling rate) | 45 | |||
G.711 (8KHz sampling rate) | 48 | |||
6 video feeds + 1 screen sharing | Yes | Opus (48KHz sampling rate) | 45 | |
Opus (8KHz sampling rate) | 48 | |||
G.722 (16KHz sampling rate) | 44 | |||
G.711 (8KHz sampling rate) | 48 | |||
9 video feeds + 1 screen sharing | No | Opus (48KHz sampling rate) | 40 | |
Opus (8KHz sampling rate) | 42 | |||
G.722 (16KHz sampling rate) | 40 | |||
G.711 (8KHz sampling rate) | 45 | |||
UCM6302 | 4 video feeds + 1 screen sharing | Yes | Opus (48KHz sampling rate) | 45 |
Opus (8KHz sampling rate) | 48 | |||
G.722 (16KHz sampling rate) | 45 | |||
G.711 (8KHz sampling rate) | 44 | |||
6 video feeds + 1 screen sharing | Yes | Opus (48KHz sampling rate) | 45 | |
Opus (8KHz sampling rate) | 48 | |||
G.722 (16KHz sampling rate) | 45 | |||
G.711 (8KHz sampling rate) | 48 | |||
9 video feeds + 1 screen sharing | No | Opus (48KHz sampling rate) | 40 | |
Opus (8KHz sampling rate) | 42 | |||
G.722 (16KHz sampling rate) | 40 | |||
G.711 (8KHz sampling rate) | 45 | |||
UCM6304 | 4 video feeds + 1 screen sharing | Yes | Opus (48KHz sampling rate) | 84 |
Opus (8KHz sampling rate) | 88 | |||
G.722 (16KHz sampling rate) | 82 | |||
G.711 (8KHz sampling rate) | 84 | |||
6 video feeds + 1 screen sharing | Yes | Opus (48KHz sampling rate) | 82 | |
Opus (8KHz sampling rate) | 86 | |||
G.722 (16KHz sampling rate) | 82 | |||
G.711 (8KHz sampling rate) | 84 | |||
9 video feeds + 1 screen sharing | No | Opus (48KHz sampling rate) | 80 | |
Opus (8KHz sampling rate) | 85 | |||
G.722 (16KHz sampling rate) | 80 | |||
G.711 (8KHz sampling rate) | 82 | |||
UCM6308 | 4 video feeds + 1 screen sharing | Yes | Opus (48KHz sampling rate) | 84 |
Opus (8KHz sampling rate) | 88 | |||
G.722 (16KHz sampling rate) | 82 | |||
G.711 (8KHz sampling rate) | 84 | |||
6 video feeds + 1 screen sharing | Yes | Opus (48KHz sampling rate) | 82 | |
Opus (8KHz sampling rate) | 86 | |||
G.722 (16KHz sampling rate) | 82 | |||
G.711 (8KHz sampling rate) | 84 | |||
9 video feeds + 1 screen sharing | No | Opus (48KHz sampling rate) | 80 | |
Opus (8KHz sampling rate) | 85 | |||
G.722 (16KHz sampling rate) | 80 | |||
G.711 (8KHz sampling rate) | 82 |
Meetings with 4 or fewer video feeds
Formula:
Required bandwidth per participant (Mbps) = (A * 1 + B * 0.172) * 1.2 + 0.1
Where:
- A = number of 1080p video feeds (A = 0 or 1)
- B = number of participants excluding A (B = total participants – A)
- 1.2 = overhead factor accounting for packet headers and other transmission costs
- 0.1 Mbps = allocated bandwidth for audio
Meetings with more than 4 video feeds
Formula:
Required bandwidth per participant (Mbps) = X * 0.172 * 1.2 + 0.1
Where:
- X = total number of participants
- 1.2 = overhead factor for packet headers and transmission overhead
- 0.1 Mbps = bandwidth for audio
Meetings with screen sharing enabled
Formula:
Required bandwidth per participant (Mbps) = (X * 0.172 + 0.512) * 1.2 + 0.1
Where:
- X = total number of participants
- 0.512 Mbps = additional bandwidth required for screen sharing
- 1.2 = overhead factor for packet headers and transmission overhead
- 0.1 Mbps = bandwidth for audio
No, only one video feed can be 1080p or 720p. Also, if there are more than 4 video feeds during a meeting, or if there is screen sharing, the video bitrate for the 1080p/720p video feed will be reduced.
- For 1080p video: 1Mbps/15fps.
- For 720p video: 512kbps/15fps.
- For 360p video, 172kbps/15fps.
- For screen sharing, 512kbps/5fps.
The above configurations can be modified per user’s preferences and it may impact the overall max number of allowed conference attendees depending on the trade-offs between video quality and max number of attendees.
In case of a power outage, the UCM630x can automatically failover its analog FXS phone ports to the corresponding PSTN FXO ports.
For example, FXS1 will connect to FXO1, allowing users to use a basic analog phone connected to the UCM to make and receive calls over the traditional phone line during the outage.
The UCM630x supports a wide range of codecs for compatibility and quality:
- Voice Codecs: Opus, G.711 (A-law/U-law), G.722, G.729, G.726, iLBC, GSM, and more.
- Video Codecs: H.264, H.263, H.263+, VP8.
Yes. The number of built-in ports can be expanded by peering a compatible FXS or FXO gateway to the UCM630x.
The UCM630x offers extensive multi-language support:
- Web User Interface: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, and Turkish.
- IVR/Voice Prompts: English, Chinese, German, Spanish, French, and many more, with the ability to add custom language packs.
Initialization & Setup
If you are using UCM RemoteConnect service, the UCM6300 series behind NAT can already work with your Grandstream end devices and Wave applications without additional NAT settings in your network routers.
However, if end users are experiencing audio issues, please navigate to UCM6300 series web UI→RemoteConnect→Plan Settings and ensure Media NAT Traversal Service is enabled, which is the default settings.
If you are not using UCM RemoteConnect service, please see configurations for different scenarios below. For more details, please refer to UCM6300 Remote Work Environment Setup Guide.
To use the same SIP extension across multiple devices, such as a deskphone, Wave Web, Wave Desktop, and Wave Mobile, your extension must allow 3 or more concurrent registrations. By default, UCM6300 series extensions are set to allow 3 concurrent registrations. To verify or adjust this setting, log in to the UCM6300 web UI as an admin and navigate to:
Extension/Trunk → Extensions → Edit Extension → Concurrent Registrations
Additionally, Wave clients support a setting called “Simultaneous Login with the Same Client Type“, which controls whether you can sign in to multiple Wave clients of the same type at the same time (for example, two desktop apps or multiple mobile devices).
- This option appears only if the UCM administrator has granted the necessary permission to the extension. If you do not see it, contact your system administrator.
- If this setting is disabled, signing in to a second Wave client of the same type will automatically sign out the first.
To register SIP extension on Grandstream end device, please refer to RemoteConnect endpoint configuration guide here.
The UCM630x supports Zero Config (auto-provisioning). Ensure the phone and UCM are on the same network. The UCM will automatically discover the phone and provision it with a chosen extension configuration, requiring minimal manual intervention.
This setting allows only one device to be registered at a time. If a new device registers with this extension, it will seize (take over) the registration from the previously registered device. This is useful for a single user switching between devices.
In the extension’s configuration under the Media tab, check the “NAT” option and set “Enable Direct Media” to “Yes”.
Typically, for remote endpoints, you should enable NAT and consider disabling Enable Direct Media to force media through the UCM, which often resolves one-way audio.
Go to Web GUI → Extension/Trunk → Extensions. The “Status” column shows the registration state for each extension:
- Green (Idle): Registered and ready.
- Blue (Ringing): Incoming call.
- Yellow (In Use): On an active call.
- Grey (Unavailable): Not registered or disabled.
Call Management & Advanced Features
You may need to enable “Allow Visitor to Call” option on the UCM Web UI→Messaging→Live Chat→Create New Live Chat→Call Settings module.
Please make sure that your have configured WebRTC Trunk with the corresponding inbound route. If the destination number configured is a call queue, please make sure that there are online agents in the call queue.
You can send the Live Chat link via email or text, or in a message on Facebook Messenger or Twitter. You can also generate a QR code and use it as needed.
Please log in UCM63xx web UI and navigate to PBX Settings→SIP Settings→Misc page, select “Enable Use of Final SDP” to ensure it’s enabled. Save and apply the setting.
The Follow Me feature allows you to define a sequence of numbers (e.g., your desk phone, then your mobile) for incoming calls to try. The system will ring each number in your designated order until the call is answered or it reaches the final destination, like voicemail.
You can use the Forwarding Exemption List (or Whitelist) in your extension’s Call Policy. Calls from numbers on this list will ring your extension normally and will not be forwarded, even if call forwarding is active.
Call Privileges (Internal, Local, National, International) control which outbound routes a user can access. A user must have a privilege level equal to or higher than the level set on an outbound route to make calls using that route. For example, a user with “Local” privileges cannot use a route set to “National”.
Yes. In the extension’s Features settings, you can set a “Maximum Call Duration”. Once this time limit (in seconds) is reached, the call will be automatically terminated.
Hot-Desking allows different users to log into the same physical phone using their own extension and password. This is useful in shared workspaces or call centers where a single phone is used by multiple people at different times.
LDAP & Phonebook Management
For external contacts on UCM6300 address book, the maximum number is 5000. Please note there is no limitation on external LDAP contacts.
UCM6300 series has no hard limit on how many LDAP contacts it can store. However, 20,000 contacts is the recommended limit as more will noticeably impact UCM web page responsiveness.
When an endpoint queries contacts from UCM LDAP server, the number of returned contacts will be limited by the number of contacts supported by the endpoint.
The UCM can act as an LDAP client. Go to System Settings→LDAP Server→LDAP Phonebook→Phonebook Download Configurations. Here, you can enter the remote server’s address, Base DN, and login credentials to pull contacts.
Yes. You can import contacts from a CSV, XLS, VCF, or XML file by clicking the “Import Phonebook” button in the LDAP Phonebook section. The file must be UTF-8 encoded.
If your phone is provisioned by the UCM via Zero Config, the LDAP settings can be pushed automatically. For manual setup, enter the UCM’s IP address as the LDAP server, port 389, and the Base DN (usually dc=pbx,dc=com).
Configuration, Provisioning, & Firmware
The UCM6300 series follows the rules below when providing the end device’s Config server URL based on UCM WAN/LAN address.
- For UCM Network Interface Method: Dual
- If the end device is in the same subnet as UCM LAN 1, Zeroconfig provisioning URL will be based on UCM LAN 1 IP.
- If the end device is in the same subnet as UCM LAN 2, Zeroconfig provisioning URL will be based on UCM LAN 2 IP.
- If the end device IP does not appear to be in the same subnet as UCM, the IP address based on the interface selected as the Default Interface under UCM webUI→System Settings→Network Settings will be used.
- For UCM Network Interface Method: Route
- If the end device is in the same subnet as UCM LAN, for example, UCM LAN IP is 192.168.2.1, end device IP is 192.168.2.100, UCM WAN IP is 10.16.254.68:
- If end device IP is in UCM Zeroconfig subnet whitelist, Zeroconfig provisioning IP will be based on UCM WAN IP.
- If end device IP is not in UCM Zeroconfig subnet whitelist, Zeroconfig provisioning IP will be based on UCM LAN IP.
- If the end device is in the same subnet as UCM WAN, for example, end device IP: 192.168.129.88, UCM WAN IP: 192.168.129.66 (they have the same gateway), Zeroconfig provisioning IP will be based on UCM WAN IP.
- If the end device IP does not appear to be in the same subnet as UCM, for example, UCM WAN IP: 192.168.129.88, UCM LAN IP: 192.168.2.1, end device IP: 10.16.254.68:
- If end device IP is in UCM Zeroconfig subnet whitelist, Zeroconfig provisioning URL will be based on UCM WAN IP.
- If end device IP is not in UCM Zeroconfig subnet whitelist, Zeroconfig provisioning URL will be based on UCM LAN IP.
- If the end device is in the same subnet as UCM LAN, for example, UCM LAN IP is 192.168.2.1, end device IP is 192.168.2.100, UCM WAN IP is 10.16.254.68:
- For UCM Network Interface Method: Switch
- The Zeroconfig provisioning URL will be based on the UCM network interface IP address. There is only one IP address under the Switch network interface method.
While you cannot directly restore UCM6300 backups to GCC, you can convert UCM6300 backups to become compatible with GCC-PBX and then restore the backups normally.
To do so, log in your GDMS account and navigate to GDMS Sidebar→UCMRC→Cloud Storage, click on the Convert Config File button in the top right corner.
This feature is supported only on UCM6300 firmware versions 1.0.27.x and above.
Uploading UCM62xx/6510 backups directly to a UCM63xx series model is not supported. However, users can convert UCM62xx/6510 backups to be UCM63xx-compatible from GDMS.
To do so, log in your GDMS account and navigate to GDMS Sidebar→UCMRC→Cloud Storage, click on the Convert Config File button in the top right corner.
This feature is only supported if the UCM62xx/6510 firmware version is 1.0.20.38 or higher.
Upgrading UCM firmware requires following a stepwise approach to ensure compatibility and avoid issues. Here’s the recommended process:
- Check your current firmware version
- Identify the version currently installed on your device (navigate to System Status→System Information→General). Devices on very old firmware may require upgrading through one or more intermediate versions.
- Read the firmware release notes
- Release notes (which can be found here) indicate version compatibility, required intermediate upgrades, feature changes, and any limitations (e.g., downgrade may not be supported).
- Backup your configuration and data
- Always perform a full backup before upgrading to prevent data loss in case of upgrade failure.
- Follow intermediate upgrade steps if required
- Some firmware versions cannot be reached directly from older versions. For example, devices below 1.0.7.x may need to upgrade first to 1.0.9.x before installing 1.0.11.x or later.
- Upgrade to the latest compatible firmware
- After completing necessary intermediate upgrades, install the latest firmware to benefit from improved stability, security, and features.
After downgrading the UCM63xx device, users need to factory reset the UCM63xx first. Otherwise, users may experience certain unexpected issues due to database errors.
Yes, through the 1+N Media Cluster feature. This advanced capability allows you to cluster multiple supported devices (UCM6304, or UCM6308) together.
One unit acts as the “Business Server” handling management, while additional “Media Servers” offload the voice and video traffic, increasing total capacity.
A single cluster can consist of one Business Server and up to five Media Servers.
The table below illustrates the scalability for the supported UCM models:
UCM6304 | UCM6308 | |
Business Server | ||
Max Concurrent Calls | 300 | 450 |
Max Meeting Participants | 200 | 300 |
Media Server | ||
Additional Concurrent Calls | 335 | 390 |
Additional Meeting Participants | 250 | 300 |
Troubleshooting & Maintenance
This is because the UCM6300 series use self-signed certificate for web server by default. To avoid this issue, you could use UCM RemoteConnect plan for the UCM6300 series, which will provide domain name for the UCM. When users access Wave Web or admin accesses UCM Web portal, the web server will trust it and allow access.
Alternatively, you can purchase a domain name and certificate from other 3rd parties. The certificate can then be uploaded to UCM6300 series web UI→System Settings→HTTP Server→Certificate Settings.
To renew your certificate, you can do either of the following:
- Obtain and upload a new certificate from your certificate authority (CA) or generate one from OpenSSL.
- For HTTPS certificates, you can also request for a new certificate in the System Settings→HTTP Server page.
To apply your new certificates, log into the UCM management portal as the super administrator and go to one of the following pages:
- For SIP TLS certificates, navigate to PBX Settings→SIP Settings→TCP/TLS.
- For HTTPS certificates, navigate to System Settings→HTTP Server→Certificate Settings.
From these pages, you can upload and apply the new certificates obtained from your certificate authorities (CA) or OpenSSL.
You can configure this under Maintenance→Syslog. Enter your remote syslog server’s IP address and select the specific modules and log levels you wish to forward.
Navigate to Maintenance→Network Troubleshooting→Ethernet Capture. Here you can:
- Select an interface (e.g., WAN, LAN).
- Apply a filter (e.g., host 192.168.1.100).
- Include the TLS handshake process to decrypt SSL/TLS packets through Wireshark.
- Start the capture, reproduce the issue, then stop and download the .pcap file for analysis.
Yes. Use the “A Key Dial-up FXO” tool under Maintenance→Signaling Troubleshooting. Enter an external phone number and click “Start” to automatically place a test call through the trunk and capture a diagnostic trace.
You can use the built-in network tools under Maintenance→Network Troubleshooting.
- Use Ping to test basic connectivity.
- Use Traceroute to identify the network path and locate where a connection may be failing.
Network & Security
The UCM6300 series PBX supports (but not limited to) below Internet Protocols Standards:
- RFC 3261: SIP: Session Initiation Protocol
- RFC 3262: Reliability of Provisional Responses in SIP
- RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers
- RFC 3264: An Offer/Answer Model with the Session Description Protocol
- RFC 3515: The Session Initiation Protocol (SIP) Refer Method
- RFC 3311: The Session Initiation Protocol (SIP) UPDATE Method
- RFC 4028: Session Timers in the Session Initiation Protocol (SIP)
- RFC 2976: The SIP INFO Method
- RFC 3842: A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)
- RFC 3892: The Session Initiation Protocol (SIP) Referred-By Mechanism
- RFC 3428: Session Initiation Protocol (SIP) Extension for Instant Messaging
- RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals
- RFC 4566: SDP: Session Description Protocol
- RFC 2617: HTTP Authentication; Basic and Digest Access Authentication
- RFC 3856: A Presence Event Package for the Session Initiation Protocol (SIP)
- RFC 3711: The Secure Real-time Transport Protocol (SRTP)
- RFC 4582: The Binary Floor Control Protocol (BFCP) (Not applicable to UCM6300A series)
- RFC 4583: Session Description Protocol (SDP) Format for Binary Floor Control Protocol (BFCP) Streams
- RFC 5245: Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols
- RFC 5389: Session Traversal Utilities for NAT (STUN)
- RFC 5766: Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)
- RFC 6347: Datagram Transport Layer Security Version 1.2
- RFC 6455: The WebSocket Protocol
- RFC 8860: Sending Multiple Types of Media in a Single RTP Session
- RFC 4734: Definition of Events for Modem, Fax, and Text Telephony Signals
- RFC 3665: Session Initiation Protocol (SIP) Basic Call Flow Examples
- RFC 3323: A Privacy Mechanism for the Session Initiation Protocol (SIP)
- RFC 3550: RTP: A Transport Protocol for Real-Time Applications
- NACK:
By enabling negative-acknowledgement (NACK), the UCM6300 series will retransmit packets that have been lost in the initial transmission to repair the media stream.
- NACK+RTX:
It’s a mechanism based on NACK, which means it relies on RTCP packets to find out which packets are lost first. For NACK, it will retransmit based on the original packet. With RTX, a special payload is used to retransmit the packets that a NACK request indicated as lost.
Retransmitted packets are sent in a different stream from the original media stream. The payload of the retransmission packet contains the payload header of the retransmission followed by the payload of the original packet. RTX retransmits using an extra ssrc, which will be marked in SDP during negotiation.
NACK+RTX can have more accurate statistics, and it’s recommended for RemoteConnect users if there is packet loss during usage.
These are options to minimize the effects of audio/video packet loss and allow the UCM6300 series to handle up to 50% packet loss for audio/video. They use the proprietary GS-FEC algorithm.
ICE is a NAT traversal method for the UCM6300 series’ Wave WebRTC functionality and UCM6300 series SIP trunk has supported it. When using UCM6300 series SIP trunk service for Wave, if ICE Support is not enabled on UCM6300 series, you may experience audio, video and screen sharing issues.
Besides enabling ICE support on the trunk of UCM6300 series, the remote side (UCM or other trunk provider service) of the trunk needs to support ICE as well.
Enabling NetEQ on the UCM6300 series helps reduce the effects of packet loss on audio received by the UCM. It is a dynamic jitter buffer and error concealment algorithm used for hiding the negative effects of network jitter and packet loss. It helps keep latency as low as possible while maintaining the highest voice quality.
Enabling NetEQ will help with minimizing the effects of packet loss on audio received by the UCM6300 series. If there is packet loss in the audio sent from UCM to an endpoint, then the endpoint will need to handle it with its own packet loss mitigation implementation.
NetEQ is used in WebRTC for audio QoS purposes. As such, UCM6300 series’ Grandstream Wave, which uses WebRTC, comes with NetEQ support.
Yes. Use the Geo-IP Access Control feature (under System Settings→Security Settings) to block all web GUI, API, and extension registration attempts from IP addresses originating in specific countries.
Navigate to System Settings→Security Settings→Static Defense. Click Create New Rule and set the Action to “DROP” or “REJECT“, set the Type to “IN“, and specify the malicious IP address in the Source IP Address field.
The Data/File Encryption feature under System Settings→Security Settings can be used to encrypt data on the UCM6300. You can select which file types (e.g., recordings, voicemails) to encrypt. The system will generate a secret key that you must export and store securely, as it is required to decrypt the files.