IP-PBX Phone Systems in Pakistan:
Experience seamless communication with ClickTech’s advanced IP-PBX Phone Systems in Pakistan. Elevate your business connectivity with our reliable and feature-rich solutions. Our IP-PBX systems streamline communication, offering crystal-clear voice quality and robust functionality. Explore cost-effective and scalable options tailored to meet the unique needs of your organization. Trust ClickTech for cutting-edge technology that ensures efficient and hassle-free communication. Upgrade to the future of business telephony with our IP-PBX Phone Systems – the key to unlocking unparalleled connectivity and productivity.
Grandstream GXW4104
- 4 or 8 ports
- 2 10/100 Mbps network ports
- Comprehensive codec support, caller ID, flexible dial plans, and security protection
- Advanced security protection with SRTP
- Designed and tested for full interoperability with leading IP-PBXs, soft-switches, and SIP-based environments
- PSTN Failover on power failure
Grandstream GXW4108
- 4 or 8 ports
- 2 10/100 Mbps network ports
- Comprehensive codec support, caller ID, flexible dial plans, and security protection
- Advanced security protection with SRTP
- Designed and tested for full interoperability with leading IP-PBXs, soft-switches, and SIP-based environments
- PSTN Failover on power failure
Grandstream GXW4200
- 16/24/32 FXS ports, GXW4248 includes 2 50-pin Telco connectors
- 1 Gigabit network port
- 132x48 backlit graphic display with support for multiple languages
- 4 SIP server profiles per system, independent SIP account per port
- Designed and tested for full interoperability with leading IP-PBXs, soft switches, and SIP-based environments
- Advanced security protection with SRTP/TLS/HTTPS
- Supports 16KHz wide-band voice sampling rate and wide-band codecs such as Opus and G.722.
- Carrier-grade security features include secure boot, unique certificate and random default password per device, and dual firmware images.
- Supports simultaneous 3-way voice conferencing per port
Grandstream GXW4500
- Software configurable E1/T1/ J1 ports, support PRI, SS7, MFC R2
- Dual Gigabit auto- sensing RJ45 network ports with integrated NAT router
- Supports a wide- range of voice codecs, including Opus, G.722, G.729, iLBC, and more
- TLS and SRTP security encryption technology to protect calls and accounts
- Automated provisioning by HTTP/TFTP with XML config files
- Supports T.38 Fax for creating Fax-over-IP
- Supports multi- language voice prompts
Grandstream HT841
- Supports 3 SIP profiles through 1 FXS port and 8 FXO ports
- High-performance NAT router
- Lifeline support (FXS port will be hard-relayed to FXO port) in case of a power outage
- 3-way voice conferencing per port
- Automated & secure provisioning options using TR069
- Supports T.38 Fax for reliable Fax-over-IP
- A Failover SIP server automatically switches to a secondary server if the main server loses connection
- Strong AES encryption with security certificate per unit
Grandstream HT881
- Supports 3 SIP profiles through 1 FXS port and 8 FXO ports
- High-performance NAT router
- Lifeline support (FXS port will be hard-relayed to FXO port) in case of a power outage
- 3-way voice conferencing per port
- Automated & secure provisioning options using TR069
- Supports T.38 Fax for reliable Fax-over-IP
- A Failover SIP server automatically switches to a secondary server if the main server loses connection
- Strong AES encryption with security certificate per unit
Grandstream UCM6300A
- Supports up to 250 users and up to 50 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
- API available for third-party integrations, including CRM and PMS platforms
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
- Automated NAT firewall traversal service facilitates secure remote connections
- Enhanced reliability with support for Hot Standby High Availability and local dual deployment
- Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
- Compatible with GDMS for cloud setup, management and monitoring
Grandstream UCM6301
- Supports up to 2000 users and 200 SIP trunk accounts
- 1GHz quad-core Cortex A9 processor
- 1GB DDR3 RAM, 32GB Flash
- 1 Integrated T1/E1/J1 interface, 2PSTN trunk FXO ports with lifeline capability
- Gigabit network ports with PoE, USB, SD card, integrated NAT router
- security protection using SRTP, TLS, and HTTPS
Grandstream UCM6302A
- Supports up to 500 users and up to 75 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
- API available for third-party integrations, including CRM and PMS platforms
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
- Automated NAT firewall traversal service facilitates secure remote connections
- Enhanced reliability with support for Hot Standby High Availability and local dual deployment
- Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
- Compatible with GDMS for cloud setup, management and monitoring
Grandstream UCM6304A
- Supports up to 1,000 users and up to 120 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
- API available for third-party integrations, including CRM and PMS platforms
- Advanced security protection with secure boot, unique certificate, and random default password to protect calls and accounts
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
- Automated NAT firewall traversal service facilitates secure remote connections
- Enhanced reliability with support for Hot Standby High Availability and local dual deployment
- Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
- Compatible with GDMS for cloud setup, management and monitoring
Grandstream UCM6308A
- Supports up to 1500 users and up to 200 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Enhanced reliability with support for Hot Standby high availability and local dual deployment
- Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
- Compatible with GDMS for cloud setup, management, and monitoring
- Based on Asterisk* version 16 open source telephony operating system
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
- Automated NAT firewall traversal service facilitates secure remote connections
- API available for third-party integrations, including CRM and PMS platforms
Grandstream UCM6510
- Supports up to 2000 users and 200 SIP trunk accounts
- 1GHz quad-core Cortex A9 processor
- 1GB DDR3 RAM, 32GB Flash
- 1 Integrated T1/E1/J1 interface, 2PSTN trunk FXO ports with lifeline capability
- Gigabit network ports with PoE, USB, SD card, integrated NAT router
- Comprehensive security protection using SRTP, TLS, and HTTPS with hardware encryption accelerator
Grandstream HA100
- Automated failover solution for the UCM6510 IP PBX, ideal for businesses that require an always-on, redundant voice system.
- Constantly monitors the operation status of both UCM6510 and automatically switches the system control to the hot-standby secondary UCM6510 in the event that the primary UCM6510 fails.
- Can complete the entire system switch between 10 and 50 seconds depending on the number of registered SIP endpoints.
- Smart monitoring and automated failover capability ensure maximum total system reliability and uptime.
- Up to 14 LED indicators showing the real-time status of all of the telecom lines, network links, auxiliary devices, etc.
- Gratuitous ARP forces SIP endpoints to refresh the MAC address of the new UCM6510 without interruptions.
- Connects and constantly monitors two UCM6510 together for high availability.
- Fast 10 to 50-second system switching time depending on the number of registered endpoints. Ideal for businesses that require a high-availability solution for the UCM6510 to ensure maximum total system reliability and uptime.
Grandstream UCM6202
- Supports up to 500 users and 30 concurrent calls
- Integrated 2/4 PSTN trunk FXO ports, 2 analog telephone FXS ports, and up to 50 SIP trunk accounts
- Gigabit network ports with integrated PoE, USB, and SD card
- Built-in call recording server and call detail records (CDR)
- 5-level IVR (Interactive Voice Response)
- Advanced security features including TLS and SRTP encryption
- Auto Discovery and Zero Configuration of Grandstream SIP endpoints
- Supports any SIP video endpoint using H.264, H.263, or H.263+ codecs
- Connects and monitors two UCM6510 together for high availability
- Smart failover solution for automatic switching to a hot-standby secondary UCM6510 if the primary one fails
Grandstream UCM6204
- Centralized solution for the communication needs of businesses
- Combines enterprise-grade voice, video, data, and mobility features in a single device
- Supports up to 500 users, 200 SIP trunk accounts, and up to 100 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Strong security protection using SRTP, TLS, and HTTPS encryption
- Supports up to a 5-level IVR (Interactive Voice Response)
- Built-in call recording server, call queue, and Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Gigabit network ports with integrated PoE, USB, and SD card
- Supports any SIP video endpoint that uses the H.264, H.263, or H.263+ codecs
- Multi-language auto-attendant and call queue to efficiently handle incoming calls
Grandstream UCM6208
- Supports up to 800 users and 100 concurrent calls
- Auto Discovery and Zero Configuration of Grandstream SIP endpoints
- Integrated 2/4/8 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability, and up to 200 SIP trunk accounts
- Gigabit network ports with integrated PoE, USB, and SD card
- Supports up to a 5-level IVR (Interactive Voice Response)
- Built-in call recording server; recordings accessible via web user interface
- Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Multi-language auto-attendant and call queue to efficiently handle incoming calls
- Strong security protection using SRTP, TLS, and HTTPS encryption
- Supports any SIP video endpoint using H.264, H.263, or H.263+ codecs
Grandstream UCM6302
- Supports up to 1000 users and up to 150 concurrent calls
- Zero-configuration provisioning of Grandstream SIP endpoints
- Built-in conferencing and meetings platform
- Advanced security protection with secure boot, unique certificate, and random default password
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support for NAT router
- Can be paired with the UCM6300 ecosystem for a hybrid platform combining on-premise IP PBX control with remote cloud access
Grandstream UCM6304
- Supports up to 800 users, 200 SIP trunk accounts, and up to 100 concurrent calls
- Enterprise-grade voice, video, and data features, including zero-configuration provisioning of
- Grandstream SIP endpoints, strong security protection using SRTP, TLS, and HTTPS encryption, and multi-language auto-attendant to efficiently handle incoming calls
- Integrated LDAP and XML phonebooks, flexible dial plan, call queue for efficient call volume management, and built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Dual Gigabit network ports with integrated PoE+ and supports any SIP video endpoint using various codecs
- Dual-core 1GHz processor, 1GB of RAM, and 4GB of flash memory
- Dimensions of 440mm L x 185mm W x 44mm H and a weight of 2.23kg
- No licensing fees or additional costs per feature required
- Ideal for small and medium-sized businesses seeking a reliable and feature-rich communication system
Grandstream UCM6308
- Supports up to 800 users and up to 100 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Strong security protection using SRTP, TLS, and HTTPS encryption
- Multi-language auto-attendant, integrated LDAP and XML phonebooks, and flexible dial plan
- Supports any SIP video endpoint using various codecs
- Offers voicemail and fax forwarding to email
- Built-in web meetings and video conferencing solution
- Can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of an on-premise IP PBX with the remote access of a cloud solution
- Provides a centralized solution for the communication needs of businesses, unifying all business communication on one centralized network
- Offers a suite of mobility, security, meeting, and collaboration tools, providing a powerful platform for any organization
IP-PBX Phone Systems in Pakistan:
Experience seamless communication with ClickTech’s advanced IP-PBX Phone Systems in Pakistan. Elevate your business connectivity with our reliable and feature-rich solutions. Our IP-PBX systems streamline communication, offering crystal-clear voice quality and robust functionality. Explore cost-effective and scalable options tailored to meet the unique needs of your organization. Trust ClickTech for cutting-edge technology that ensures efficient and hassle-free communication. Upgrade to the future of business telephony with our IP-PBX Phone Systems – the key to unlocking unparalleled connectivity and productivity.
Grandstream GXW4104
- 4 or 8 ports
- 2 10/100 Mbps network ports
- Comprehensive codec support, caller ID, flexible dial plans, and security protection
- Advanced security protection with SRTP
- Designed and tested for full interoperability with leading IP-PBXs, soft-switches, and SIP-based environments
- PSTN Failover on power failure
Grandstream GXW4108
- 4 or 8 ports
- 2 10/100 Mbps network ports
- Comprehensive codec support, caller ID, flexible dial plans, and security protection
- Advanced security protection with SRTP
- Designed and tested for full interoperability with leading IP-PBXs, soft-switches, and SIP-based environments
- PSTN Failover on power failure
Grandstream GXW4200
- 16/24/32 FXS ports, GXW4248 includes 2 50-pin Telco connectors
- 1 Gigabit network port
- 132x48 backlit graphic display with support for multiple languages
- 4 SIP server profiles per system, independent SIP account per port
- Designed and tested for full interoperability with leading IP-PBXs, soft switches, and SIP-based environments
- Advanced security protection with SRTP/TLS/HTTPS
- Supports 16KHz wide-band voice sampling rate and wide-band codecs such as Opus and G.722.
- Carrier-grade security features include secure boot, unique certificate and random default password per device, and dual firmware images.
- Supports simultaneous 3-way voice conferencing per port
Grandstream GXW4500
- Software configurable E1/T1/ J1 ports, support PRI, SS7, MFC R2
- Dual Gigabit auto- sensing RJ45 network ports with integrated NAT router
- Supports a wide- range of voice codecs, including Opus, G.722, G.729, iLBC, and more
- TLS and SRTP security encryption technology to protect calls and accounts
- Automated provisioning by HTTP/TFTP with XML config files
- Supports T.38 Fax for creating Fax-over-IP
- Supports multi- language voice prompts
Grandstream HT841
- Supports 3 SIP profiles through 1 FXS port and 8 FXO ports
- High-performance NAT router
- Lifeline support (FXS port will be hard-relayed to FXO port) in case of a power outage
- 3-way voice conferencing per port
- Automated & secure provisioning options using TR069
- Supports T.38 Fax for reliable Fax-over-IP
- A Failover SIP server automatically switches to a secondary server if the main server loses connection
- Strong AES encryption with security certificate per unit
Grandstream HT881
- Supports 3 SIP profiles through 1 FXS port and 8 FXO ports
- High-performance NAT router
- Lifeline support (FXS port will be hard-relayed to FXO port) in case of a power outage
- 3-way voice conferencing per port
- Automated & secure provisioning options using TR069
- Supports T.38 Fax for reliable Fax-over-IP
- A Failover SIP server automatically switches to a secondary server if the main server loses connection
- Strong AES encryption with security certificate per unit
Grandstream UCM6300A
- Supports up to 250 users and up to 50 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
- API available for third-party integrations, including CRM and PMS platforms
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
- Automated NAT firewall traversal service facilitates secure remote connections
- Enhanced reliability with support for Hot Standby High Availability and local dual deployment
- Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
- Compatible with GDMS for cloud setup, management and monitoring
Grandstream UCM6301
- Supports up to 2000 users and 200 SIP trunk accounts
- 1GHz quad-core Cortex A9 processor
- 1GB DDR3 RAM, 32GB Flash
- 1 Integrated T1/E1/J1 interface, 2PSTN trunk FXO ports with lifeline capability
- Gigabit network ports with PoE, USB, SD card, integrated NAT router
- security protection using SRTP, TLS, and HTTPS
Grandstream UCM6302A
- Supports up to 500 users and up to 75 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
- API available for third-party integrations, including CRM and PMS platforms
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
- Automated NAT firewall traversal service facilitates secure remote connections
- Enhanced reliability with support for Hot Standby High Availability and local dual deployment
- Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
- Compatible with GDMS for cloud setup, management and monitoring
Grandstream UCM6304A
- Supports up to 1,000 users and up to 120 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
- API available for third-party integrations, including CRM and PMS platforms
- Advanced security protection with secure boot, unique certificate, and random default password to protect calls and accounts
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
- Automated NAT firewall traversal service facilitates secure remote connections
- Enhanced reliability with support for Hot Standby High Availability and local dual deployment
- Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
- Compatible with GDMS for cloud setup, management and monitoring
Grandstream UCM6308A
- Supports up to 1500 users and up to 200 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Enhanced reliability with support for Hot Standby high availability and local dual deployment
- Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
- Compatible with GDMS for cloud setup, management, and monitoring
- Based on Asterisk* version 16 open source telephony operating system
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
- Automated NAT firewall traversal service facilitates secure remote connections
- API available for third-party integrations, including CRM and PMS platforms
Grandstream UCM6510
- Supports up to 2000 users and 200 SIP trunk accounts
- 1GHz quad-core Cortex A9 processor
- 1GB DDR3 RAM, 32GB Flash
- 1 Integrated T1/E1/J1 interface, 2PSTN trunk FXO ports with lifeline capability
- Gigabit network ports with PoE, USB, SD card, integrated NAT router
- Comprehensive security protection using SRTP, TLS, and HTTPS with hardware encryption accelerator
Grandstream HA100
- Automated failover solution for the UCM6510 IP PBX, ideal for businesses that require an always-on, redundant voice system.
- Constantly monitors the operation status of both UCM6510 and automatically switches the system control to the hot-standby secondary UCM6510 in the event that the primary UCM6510 fails.
- Can complete the entire system switch between 10 and 50 seconds depending on the number of registered SIP endpoints.
- Smart monitoring and automated failover capability ensure maximum total system reliability and uptime.
- Up to 14 LED indicators showing the real-time status of all of the telecom lines, network links, auxiliary devices, etc.
- Gratuitous ARP forces SIP endpoints to refresh the MAC address of the new UCM6510 without interruptions.
- Connects and constantly monitors two UCM6510 together for high availability.
- Fast 10 to 50-second system switching time depending on the number of registered endpoints. Ideal for businesses that require a high-availability solution for the UCM6510 to ensure maximum total system reliability and uptime.
Grandstream UCM6202
- Supports up to 500 users and 30 concurrent calls
- Integrated 2/4 PSTN trunk FXO ports, 2 analog telephone FXS ports, and up to 50 SIP trunk accounts
- Gigabit network ports with integrated PoE, USB, and SD card
- Built-in call recording server and call detail records (CDR)
- 5-level IVR (Interactive Voice Response)
- Advanced security features including TLS and SRTP encryption
- Auto Discovery and Zero Configuration of Grandstream SIP endpoints
- Supports any SIP video endpoint using H.264, H.263, or H.263+ codecs
- Connects and monitors two UCM6510 together for high availability
- Smart failover solution for automatic switching to a hot-standby secondary UCM6510 if the primary one fails
Grandstream UCM6204
- Centralized solution for the communication needs of businesses
- Combines enterprise-grade voice, video, data, and mobility features in a single device
- Supports up to 500 users, 200 SIP trunk accounts, and up to 100 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Strong security protection using SRTP, TLS, and HTTPS encryption
- Supports up to a 5-level IVR (Interactive Voice Response)
- Built-in call recording server, call queue, and Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Gigabit network ports with integrated PoE, USB, and SD card
- Supports any SIP video endpoint that uses the H.264, H.263, or H.263+ codecs
- Multi-language auto-attendant and call queue to efficiently handle incoming calls
Grandstream UCM6208
- Supports up to 800 users and 100 concurrent calls
- Auto Discovery and Zero Configuration of Grandstream SIP endpoints
- Integrated 2/4/8 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability, and up to 200 SIP trunk accounts
- Gigabit network ports with integrated PoE, USB, and SD card
- Supports up to a 5-level IVR (Interactive Voice Response)
- Built-in call recording server; recordings accessible via web user interface
- Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Multi-language auto-attendant and call queue to efficiently handle incoming calls
- Strong security protection using SRTP, TLS, and HTTPS encryption
- Supports any SIP video endpoint using H.264, H.263, or H.263+ codecs
Grandstream UCM6302
- Supports up to 1000 users and up to 150 concurrent calls
- Zero-configuration provisioning of Grandstream SIP endpoints
- Built-in conferencing and meetings platform
- Advanced security protection with secure boot, unique certificate, and random default password
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support for NAT router
- Can be paired with the UCM6300 ecosystem for a hybrid platform combining on-premise IP PBX control with remote cloud access
Grandstream UCM6304
- Supports up to 800 users, 200 SIP trunk accounts, and up to 100 concurrent calls
- Enterprise-grade voice, video, and data features, including zero-configuration provisioning of
- Grandstream SIP endpoints, strong security protection using SRTP, TLS, and HTTPS encryption, and multi-language auto-attendant to efficiently handle incoming calls
- Integrated LDAP and XML phonebooks, flexible dial plan, call queue for efficient call volume management, and built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Dual Gigabit network ports with integrated PoE+ and supports any SIP video endpoint using various codecs
- Dual-core 1GHz processor, 1GB of RAM, and 4GB of flash memory
- Dimensions of 440mm L x 185mm W x 44mm H and a weight of 2.23kg
- No licensing fees or additional costs per feature required
- Ideal for small and medium-sized businesses seeking a reliable and feature-rich communication system
Grandstream UCM6308
- Supports up to 800 users and up to 100 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Strong security protection using SRTP, TLS, and HTTPS encryption
- Multi-language auto-attendant, integrated LDAP and XML phonebooks, and flexible dial plan
- Supports any SIP video endpoint using various codecs
- Offers voicemail and fax forwarding to email
- Built-in web meetings and video conferencing solution
- Can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of an on-premise IP PBX with the remote access of a cloud solution
- Provides a centralized solution for the communication needs of businesses, unifying all business communication on one centralized network
- Offers a suite of mobility, security, meeting, and collaboration tools, providing a powerful platform for any organization