How to troubleshoot no calls in or out on my NEC PBX?
If your NEC PBX system is unable to make or receive calls, first check the router configuration and the NEC PBX configuration. Also, note the WAN IP address and contact your provider to ensure it is qualified.
Why are my calls dropping on my NEC PBX?
If your NEC PBX calls are dropping, it could be due to internet access issues, extreme latency on the LAN, or a SIP issue. Contact your internet access provider and check the latency on your LAN. You may also need to contact your provider regarding the SIP issue.
What to do if my NEC PBX has no calls out?
If you are experiencing issues with making outgoing calls on your NEC PBX, first verify the NEC PBX configuration. Additionally, note the WAN IP address and contact your provider to ensure it is qualified.
What to do if my NEC PBX has no calls in?
If you are not receiving incoming calls on your NEC PBX, start by checking the NEC PBX configuration. Also, note the WAN IP address and contact your provider to ensure it is qualified.
Why are parts of words missing on my NEC PBX calls?
If you're experiencing missing parts of words on your NEC PBX calls, it could be due to packet loss or latency on the LAN or WAN. Check the LAN and consult with your internet access provider to check the WAN. Also, check the Jitter Buffer Configuration with NEC.
How to fix echo on NEC PBX?
If you are experiencing echo on your NEC PBX calls, it may be due to excessive delay or an echo cancellation issue. Check the LAN and WAN for high latency. Also, check the echo settings and/or consult IDT/Net2Phone.
Why is there static or hum on my NEC PBX phones?
If you are hearing static or hum on your NEC PBX phones, it could be a power issue. If using AC power, check the power supply. This should not be an issue if using PoE.
General| VoIP Support | Yes |
|---|---|
| VoIP Protocols | SIP, H.323 |
| Voicemail | Integrated voicemail |
| Networking | Yes, via IP |
| Unified Communications | Yes |
| Mobility | Supports mobile extensions and applications |
| Type | IP-PBX |
| Network Interfaces | Gigabit Ethernet |
| Power Supply | AC 100-240V, 50/60Hz |
| Operating Temperature | 32°F to 104°F (0°C to 40°C) |
| Storage Temperature | -20°C to 60°C (-4°F to 140°F) |
| Weight | Varies depending on chassis and expansion cards |
Provides an overview of the SV9100 K-CCIS system, including circuit cards, capacity, and network considerations.
Describes the system configuration of K-CCIS provided using a digital network.
Details the specifications and characteristics applicable to the GCD-CCTA.
Explains how to determine system configurations and supported network topologies for K-CCIS.
Covers the procedures for installing hardware components, including precautions and blade installation.
Provides essential preinstallation planning and safety precautions for installing blades.
Details the process of installing extension and trunk blades, including order of installation.
Lists programs required for K-CCIS, including digital trunk, CCIS, and numbering plan assignments.
Details the programs necessary for K-CCIS functionality, depending on features used.
Provides an overview of K-CCIS features like Automatic Recall, Call Forwarding, and others.
Lists terminal types and required components for the K-CCIS feature.
Outlines general and restricted service conditions for K-CCIS, including numbering plan details.
Lists features related to K-CCIS, such as T1 Connections and Universal Slots.
Describes the IP K-CCIS application using the GPZ-IPLE, including its role in connecting systems.
Details the data programs used when installing the GPZ-IPLE daughter board for SV9100 IP (K-CCIS).
Provides programming data assignments for a 4-digit Closed Numbering Plan using Closed Number Blocks.
Lists the port numbers for supported IP applications like SIP Trunk and H.323 Trunk.
Explains how IP K-CCIS supports Peer-to-Peer calls between IP Terminals in different offices.
Discusses various service conditions and restrictions related to IP K-CCIS and VPN implementation.
Provides an overview of UNIVERGE SV9100 IP Telephony and Voice over IP (VoIP) technology.
Describes the SV9100 as an enterprise IP Telephony solution for voice and data network convergence.
Explains Voice over IP (VoIP) technology, its use of TCP/IP, and bandwidth considerations.
Covers IP Networking concepts, including connecting systems and types of IP Networking.
Introduces IP Networking and its use of VoIP technology to connect telephone systems.
Details the SV9100 IP K-CCIS package for seamless connection of multiple systems using VoIP.
Explains SIP Trunks networking and advanced features, recommending IP K-CCIS when needed.
Illustrates typical network configurations for IP trunks, including static and dynamic IP addressing.
Describes the configuration of DTMF relay for CCISoIP networks to detect and regenerate DTMF tones.
Introduces the process of connecting the SV9100 to a data network and configuring TCP/IP.
Explains the necessity of obtaining IP addressing information and types of IP addresses.
Illustrates typical network configurations using static and dynamic IP addressing.
Describes how to test the network connection using the ICMP Ping command.
Provides detailed information on SV9100 program blocks for data network connectivity and VoIP configuration.
Guides the user on how to access the programming mode of the system via telephone.
Explains the procedure to exit programming mode and save system data, including USB backup.
Describes the use of keys for data entry, editing, and navigation within the programming menus.
Explains how to enter text for various programs using the keypad digits and softkeys.
Details the use of interactive softkeys for intuitive feature access in system programming.
Explains the meaning of various Softkey options displayed in programming mode for easy selection.
Describes the programs used to connect the SV9100 to a data network and configure VoIP functions.
Explains issues to consider when planning a UNIVERGE SV9100 VoIP installation.
Discusses Quality of Service (QoS) factors affecting VoIP speech quality and network mechanisms.
Covers issues for Voice over IP applications using internet-based connections like xDSL and cable modems.
Explains how network security measures like Firewalls and NAT can create problems for VoIP.
Describes CODEC and bandwidth and their application with the UNIVERGE SV9100 system.
Explains how to calculate the number of DSP resources required for VoIP and IP K-CCIS.
Provides an overview to classify voice traffic and implement QoS for prioritization on networks.
Lists the port numbers for supported IP applications like SIP Trunk and H.323 Trunk.
Explains the SV9100 IP Trunk SIP package for sending real-time voice over LAN or WAN using Internet protocol.
Details how the PBX supports SIP configurations using the E.164 international numbering scheme.
Describes how incoming SIP calls from external ITSPs are presented with international access code.
Introduces H.323 as an ITU standard for Packet Based Multimedia Communication Systems.
Explains the implementation and programming for H.323 trunks, including call routing and gatekeeper configuration.
Provides programming details for connecting three systems via IP CCH and one via legacy T1 K-CCIS.
Introduces IP extensions using multiline terminals connected via IP, offering similar functionality to traditional phones.
Explains how IP telephones convert speech to TDM technology using the GPZ-IPLE daughter board.
Describes the power fail adapter as an add-on module for IP and digital multiline telephones.
Details the connection of the power fail adapter to an analog PSTN line for speech path redirection.
Explains how IP telephones connect to a data network via RJ-45 ports, allowing PC sharing.
Describes methods for providing power to IP telephones, including local power and Power Over Ethernet (PoE).
Explains IP telephone communication without DSP resources for Intercom calls.
Covers system tones, ring tones, and music on hold for IP phones.
Provides typical network configurations for IP phones using static and dynamic IP addressing.
Details how to access the programming interface for IP phones via the User Menu.
Explains the use of external or internal DHCP servers for configuring IP terminal options.
Describes the required equipment and procedures for automatically configuring IP terminals.
Covers the process of registering and deleting IP phones from the SV9100 system.
Discusses considerations for using Analog Trunks with IP Phones, including echo and padding.
Explains manual and automatic firmware upgrade procedures for IP Phones.
Introduces SIP Station features for Voice over Internet Protocol (VoIP) for IP stations.
Explains how Standard SIP Terminals can initiate a conference call with SV9100 software 2.00 or higher.
Describes how any standard SIP terminal can perform an Unattended (Blind/Unsupervised) transfer.
Details the SV9100's support for NAT for Standard SIP Terminals and related conditions.
Covers basic SIP extension programming, including configuration for VoIPDB networking and IP K-CCIS.
Introduces NetLink Networking, providing seamless connection of multiple SV9100 servers.
Details hardware, capacity, license, protocol, and system ID requirements for SV9100 NetLink.
Covers NetLink connection, including IPLE card addressing and system reset procedures.
Outlines limitations for NetLink features, including synchronous ringing and IP terminal support.
Lists features supported in secondary systems, including call handling and system management.
Covers programming for NetLink functions, including system settings and port assignments.
Explains the Fail-Over feature where a secondary system takes over if the primary system is disconnected.
Describes how to determine and set the primary system for NetLink, including fail-over scenarios.
Explains database synchronization between primary and secondary systems in a NetLink network.
Recommends creating a recovery database for communication servers before disconnecting from NetLink.
Details NetLink license requirements, including activation codes and registration in the primary system.
Explains NetLink's MAC address authorization to prevent connections from unauthenticated servers.
Covers procedures for confirming NetLink basic information and communication server status.
Explains how port numbers for secondary systems are assigned based on connection order.
Describes how the primary system controls slots in secondary systems and virtual slot functionality.
Explains how to adjust the clock display by defining the time zone offset from the primary system.
Introduces NAPT (Network Address Port Translation) for private IP address translation to a public IP address.
Lists conditions for NAPT support, including firmware requirements and terminal compatibility.
Details restrictions for static NAT, including IP address assignment and router configuration.
Outlines the minimum programming required for NAPT feature implementation.
Explains the All DSP Busy feature for alerting users about DSP resource usage.
Describes service conditions for alarms related to IP Phones and SIP trunk calls.
Lists features related to All DSP Busy Indication, such as IP Multiline Station and K-CCIS.
Provides a guide to programming features related to All DSP Busy Indication.
Defines AspireNet package as a seamless connection of multiple systems using ISDN and VoIP.
Lists the available PCBs for AspireNet connection, including ISDN and VoIP options.
Lists features supported by AspireNet, such as ARS/F-Route, Barge In, and Call Forwarding.
Explains the structure of the manual, guiding users through sections on Introduction, Setup, Features, and Programming.
Discusses simplifying keyset operation with one-touch keys and programmable keys.
Explains that the type of programming depends on the selected ISDN PRIA, BRIA, or IPLE Blade.
Details AspireNet ISDN features, including B-channel ports and PRI/BRI circuit requirements.
Covers AspireNet IP, including simultaneous call limitations and CODEC type selection.
Describes AspireNet Multi Site networking of three or more systems via ISDN or IP connections.
Explains system numbering plans for routing calls, including unified numbering and flexible routing.
Details how dialed digits are sent to F-Route tables and analyzed for call routing.
Explains the Barge In feature and its options for conversation monitoring and call intervention.
Describes Call Forwarding options like Immediate, Busy, No Answer, and Both Ring with networking.
Explains how extension users can override Call Forwarding or Do Not Disturb settings on networked extensions.
Allows an extension user to forward calls to an off-site location.
Enables Call Forward Follow-Me to an extension in a networked system, causing both extensions to ring.
Allows users to call a busy extension and wait in line (Camp-On) for it to become free.
Allows Caller ID information to be sent to a target extension in a networked system.
Describes methods for seizing a trunk in a networked system for outgoing calls.
Allows users to create a conference call including users in a networked system.
To Next Page