CISCO Refurbished
- Manufacturer
- CISCO
- Condition
- Refurbished
- Lead Time
- 2-3 Days
- Availability
- Not In Stock
- List Price/RRP
- US$795.00
Description
The CISCO VG202 is a voice gateway that allows for communication between traditional analog telephones and digital voice networks. It has two voice ports that can be used to connect phones, faxes, or other analog devices.The gateway supports a range of voice codecs and has built-in security features to ensure safe communication. It also supports quality of service (QoS) and can prioritize voice traffic to ensure clear audio quality.The VG202 is easy to set up and manage, with a web-based interface for configuration and status monitoring. It is ideal for small to medium-sized businesses looking to integrate traditional telephony systems with digital networks.Overall, the CISCO VG202 provides a reliable and cost-effective solution for businesses looking to modernize their communication systems without completely abandoning their existing analog devices.- Part Code
- VG202
- Description
- Cisco VG202 Analog Voice Gateway
- Weight
- 3kg
- Manufacturer
- CISCO
- End of Hardware Support
- Yes (30 November 2018)
- End of Sale
- Yes (16 November 2013)
- End of Life
- Yes (18 May 2013)
IP Phone Support per Platform
Expand
IP Phone Support per Platform
Platform
Maximum Number of Phones
Cisco 881 and 887VA Integrated Services Routers
5
Cisco 2901 Integrated Services Router
35
Cisco 2911 Integrated Services Router
50
Cisco 2921 Integrated Services Router
100
Cisco 2951 Integrated Services Router
150
Cisco 3925 Integrated Services Router
250
Cisco 3945 Integrated Services Router
350
Cisco 3925E Integrated Services Router
400
Cisco 3945E Integrated Services Router
450
Cisco 4321 Integrated Services Router
50
Cisco 4331 Integrated Services Router
100
Cisco 4351 Integrated Services Router
250
Cisco 4431 Integrated Services Router
350
Cisco 4451-X Integrated Services Router
450
Cisco Endpoints Supported by Cisco Unified Communications Manager Express
Expand
Cisco Endpoints Supported by Cisco Unified Communications Manager Express
IP Phone Series
IP Phone Series
Models Supported
Cisco IP Phone 7800 Series
Cisco Unified IP Phone 7811
Cisco Unified IP Phone 7821
Cisco Unified IP Phone 7841
Cisco Unified IP Phone 7861
Cisco Unified IP Phone 7821
Cisco Unified IP Phone 7841
Cisco Unified IP Phone 7861
Cisco IP Phone 8800 Series
Cisco IP Phone 8811
Cisco IP Phone 8821
Cisco IP Phone 8841
Cisco IP Phone 8851
Cisco IP Phone 8861
Cisco Unified IP Conference Phone 8831
Cisco IP Phone 8821
Cisco IP Phone 8841
Cisco IP Phone 8851
Cisco IP Phone 8861
Cisco Unified IP Conference Phone 8831
Cisco Unified IP Phone 8900 Series
Cisco Unified IP Phone 8961
Cisco Unified IP Phone 8941
Cisco Unified IP Phone 8945
Cisco Unified IP Phone 8941
Cisco Unified IP Phone 8945
Cisco Unified IP Phone 9900 Series
Cisco Unified IP Phone 9951
Cisco Unified IP Phone 9971
Cisco Unified IP Phone 9971
Cisco Unified SIP Phone 3900 Series
Cisco Unified IP Phone 3905
Cisco Unified IP Phone 3911
Cisco Unified IP Phone 3911
Cisco Unified IP Phone 7900 Series
Cisco Unified IP Phone 7942G
Cisco Unified IP Phone 7945G
Cisco Unified IP Phone 7962G
Cisco Unified IP Phone 7965G
Cisco Unified IP Phone 7975G
Cisco Unified IP Phone Expansion Module 7915 and 7916 modules
Cisco Unified IP Wireless IP Phone 7925G and 7926G models
Cisco Unified IP Phone 7945G
Cisco Unified IP Phone 7962G
Cisco Unified IP Phone 7965G
Cisco Unified IP Phone 7975G
Cisco Unified IP Phone Expansion Module 7915 and 7916 modules
Cisco Unified IP Wireless IP Phone 7925G and 7926G models
Cisco DX Series
Cisco DX650
Cisco Jabber platform
Cisco Jabber for MAC and Windows
Cisco Unified Communications Manager Express 11.0 Features
Expand
Cisco Unified Communications Manager Express 11.0 Features
Feature
Phone features
Maximum 450 phones per system
Up to 34 line appearances per phone
Attendant console functions using Cisco Unified IP Phone Expansion Module 7915 and 7916 modules
Fast transfer: Blind or consult
Busy Lamp Field (BLF)
Silent ringing options
Distinctive ring per line
Automatic line selection for outbound calls
Call forward on busy, no answer, and all (internal or external)
Call-forward-all restriction control
Do not disturb (DND)
Feature ring with DND set
IP phone display of DND state
Dial-plan pattern load on SIP phones
Diversion of calls directly to voicemail
Customization of soft keys
Enable and disable call-waiting notification per line
Call waiting with overlay directory number
Call-waiting ring
Dual or eight call line appearances per button
After-hours toll-bar override
Auto answer with headset
European date formats
Hook flash pass-through across analog PSTN trunks
Idle URL: Periodically push messages or graphics on IP phones
Last-number redial
Live record to Cisco Unity Express mailbox
Local name directory lookup
On-hook dialing
Station speed dial with configuration changes from IP phone
System speed dial for 10,000 numbers
Silent and feature ring options
SIP-based line-side subscribe, providing basic presence of phone status
Transfer to voicemail soft key
Call barge with privacy on shared lines
Access features using soft keys or feature access codes
Remote teleworker IP phone support
Dynamic hunt-group join or leave
Support for analog phones using Cisco ATA 186 Analog Telephone Adaptor or Cisco VG224 Analog Voice Gateway in SCCP mode
Support for fax machines on foreign-exchange-station (FXS) ports or ATA using H.323, SCCP, or SIP
Support for Cisco VG202 and VG204 Analog Voice Gateways
XML application services on Cisco Unified IP display phones
Station-to-station video with voice using Cisco Unified Video Advantage or Cisco Unified IP Phone 7985G endpoints
Extension mobility within the single site
Wideband audio (G.722) and Internet low bit rate codec (iLBC)
Single number reach for mobility in Cisco IOS Software
Whisper intercom
Shared line on SIP phones with privacy and barge-in capabilities
Customizable phone telephony user interface (TUI) with button layout control
Improved security with customizable services and directories page
Programmable line keys (PLK) on Cisco IP Phone TUI
My Phone Apps support for BLF speed-dial configuration
Normalized +E.164 support for Cisco Unified IP Phones (SCCP)
Support for Cisco Mobile 8.1 for iPhone and iPod touch soft-phone client
Secure Sockets Layer (SSL) VPN client support for Cisco IOS Datagram Transport Layer Security (DTLS) VPN for SCCP phones
Localization support for Cisco Unified IP Phone 7920 and 6900 models with SCCP
Video and camera support for Cisco Unified IP Phone 8961, 9951, and 9971 models
Extension Mobility support for SIP endpoints
Support for Cisco Unified IP Phone 8941 and 8945 endpoints
Support for Single Number Reach and paging for SIP endpoints
Support for shared line between SCCP and SIP endpoints
My Phone App support for SIP endpoints
Support for Cisco ATA 187, ATA 188 and ATA 190 Analog Telephone Adapters
Support for SIP-initiated conference
Support for visual list of parked calls
Support for visual list of voice hunt groups
Support for dynamic hunt-group login
Support for distinctive ring for parked call recall
Support for audible ringtone with hunt-group login and logout
Up to 34 line appearances per phone
Attendant console functions using Cisco Unified IP Phone Expansion Module 7915 and 7916 modules
Fast transfer: Blind or consult
Busy Lamp Field (BLF)
Silent ringing options
Distinctive ring per line
Automatic line selection for outbound calls
Call forward on busy, no answer, and all (internal or external)
Call-forward-all restriction control
Do not disturb (DND)
Feature ring with DND set
IP phone display of DND state
Dial-plan pattern load on SIP phones
Diversion of calls directly to voicemail
Customization of soft keys
Enable and disable call-waiting notification per line
Call waiting with overlay directory number
Call-waiting ring
Dual or eight call line appearances per button
After-hours toll-bar override
Auto answer with headset
European date formats
Hook flash pass-through across analog PSTN trunks
Idle URL: Periodically push messages or graphics on IP phones
Last-number redial
Live record to Cisco Unity Express mailbox
Local name directory lookup
On-hook dialing
Station speed dial with configuration changes from IP phone
System speed dial for 10,000 numbers
Silent and feature ring options
SIP-based line-side subscribe, providing basic presence of phone status
Transfer to voicemail soft key
Call barge with privacy on shared lines
Access features using soft keys or feature access codes
Remote teleworker IP phone support
Dynamic hunt-group join or leave
Support for analog phones using Cisco ATA 186 Analog Telephone Adaptor or Cisco VG224 Analog Voice Gateway in SCCP mode
Support for fax machines on foreign-exchange-station (FXS) ports or ATA using H.323, SCCP, or SIP
Support for Cisco VG202 and VG204 Analog Voice Gateways
XML application services on Cisco Unified IP display phones
Station-to-station video with voice using Cisco Unified Video Advantage or Cisco Unified IP Phone 7985G endpoints
Extension mobility within the single site
Wideband audio (G.722) and Internet low bit rate codec (iLBC)
Single number reach for mobility in Cisco IOS Software
Whisper intercom
Shared line on SIP phones with privacy and barge-in capabilities
Customizable phone telephony user interface (TUI) with button layout control
Improved security with customizable services and directories page
Programmable line keys (PLK) on Cisco IP Phone TUI
My Phone Apps support for BLF speed-dial configuration
Normalized +E.164 support for Cisco Unified IP Phones (SCCP)
Support for Cisco Mobile 8.1 for iPhone and iPod touch soft-phone client
Secure Sockets Layer (SSL) VPN client support for Cisco IOS Datagram Transport Layer Security (DTLS) VPN for SCCP phones
Localization support for Cisco Unified IP Phone 7920 and 6900 models with SCCP
Video and camera support for Cisco Unified IP Phone 8961, 9951, and 9971 models
Extension Mobility support for SIP endpoints
Support for Cisco Unified IP Phone 8941 and 8945 endpoints
Support for Single Number Reach and paging for SIP endpoints
Support for shared line between SCCP and SIP endpoints
My Phone App support for SIP endpoints
Support for Cisco ATA 187, ATA 188 and ATA 190 Analog Telephone Adapters
Support for SIP-initiated conference
Support for visual list of parked calls
Support for visual list of voice hunt groups
Support for dynamic hunt-group login
Support for distinctive ring for parked call recall
Support for audible ringtone with hunt-group login and logout
Trunk features
Analog foreign-exchange-office (FXO) Loop and Ground Start
Ear and mouth (E&M)
Basic Rate Interface (BRI) and Primary Rate Interface (PRI) support (NI2, 4ESS, 5ESS, EuroISDN, DMS100, and DMS250) and several other switch types currently supported in Cisco IOS Software
Caller ID name and number
Automatic number identification (ANI)
Digital trunk support (T1/E1)
Direct inward dialing (DID)
Direct outward dialing (DOD)
E1 R2 support
Dedicated trunk mapping to phone button
H.323 trunks with H.450 support
H450.12 automatic detection of H.450 support for remote H.323 endpoints
H.323-to-H.323 hairpin call routing for non-H.450-compliant H.323 endpoints
SIP trunks and RFC 2833 support
Transcoding with G.711, G.729a, and iLBC
Call forwarding busy, no answer, and all
Calling line identification presentation (CLIP) and calling name identification presentation (CNIP)
Connected line identification presentation (COLP) and connected name identification presentation (CONP)
Message-waiting indicator (MWI) and message center support
MWI pass-through QSIG-to-time-division multiplexing (TDM) voicemail
Overlap sending support on ISDN PRI and BRI trunks
Ear and mouth (E&M)
Basic Rate Interface (BRI) and Primary Rate Interface (PRI) support (NI2, 4ESS, 5ESS, EuroISDN, DMS100, and DMS250) and several other switch types currently supported in Cisco IOS Software
Caller ID name and number
Automatic number identification (ANI)
Digital trunk support (T1/E1)
Direct inward dialing (DID)
Direct outward dialing (DOD)
E1 R2 support
Dedicated trunk mapping to phone button
H.323 trunks with H.450 support
H450.12 automatic detection of H.450 support for remote H.323 endpoints
H.323-to-H.323 hairpin call routing for non-H.450-compliant H.323 endpoints
SIP trunks and RFC 2833 support
Transcoding with G.711, G.729a, and iLBC
Call forwarding busy, no answer, and all
Calling line identification presentation (CLIP) and calling name identification presentation (CNIP)
Connected line identification presentation (COLP) and connected name identification presentation (CONP)
Message-waiting indicator (MWI) and message center support
MWI pass-through QSIG-to-time-division multiplexing (TDM) voicemail
Overlap sending support on ISDN PRI and BRI trunks
System features
Account codes and call-detail-record (CDR) field entry
Call-back busy subscriber and camp-on
Per-phone call-coverage rules
Call hold and retrieve
Call park: Personal and directed
Call transfer and park recall
Call park assign to extension
Call pickup directed
Call pickup local group
Call pickup explicit group
Call transfer: Consultative and blind
Call waiting
Cancel call waiting soft key
Call park and pickup on both SCCP and SIP phones
CTI with Microsoft CRM and Outlook using Cisco IOS Software Telephony Services Provider (TSP)
E911 with two emergency location numbers per zone; unlimited zones per site
Eight-party impromptu conferencing
Directory services using XML
Hunt groups: Sequential, circular, parallel (blast), and longest idle
Hunt-group dynamic log in and log out
Hunt-groups statistics: Daily and hourly
Intercom
Ad-hoc conferencing (8 parties)
Meet-me conferencing (32 parties)
Five music-on-hold (MoH) streams (internal)
Night service bell or call forwarding
Overlay extensions for enhanced call coverage
Called-name display for overlay extensions
Paging: Internal through IP phones or to external system
Per-call caller ID blocking
Secondary dial tone
Standards-based network call transfer and call forwarding using H.450
Additional system speed-dial option through XML service
Time-of-day and day-of-week call blocking
Customizable called-name display
Support of SRST fallback service phone autoregistration
Basic automatic call distributor (BACD) (three queues) with Automated Attendant and call statistics
Display of number of calls in queue on IP phone
Agent log in and log out of BACD hunt group
Integration with Cisco Unified Contact Center Express 5.0 for advanced call-center features with support for up to 50 agents, agent supervisors, call recording, silent monitoring, and reporting features
Secure Real-Time Transport Protocol (SRTP) providing media encryption for calls on the IP network
Secure voice IP phone certificate authentication and provisioning plus secure device signaling using Transport Layer Security (TLS)
Video over SIP trunk with H.264 codec support
Multilevel Precedence and Preemption (MLPP) support over PRI trunks
Enhanced MLPP with support for supplementary services, including three-way conference, call park, and call pickup
BLF monitoring of conference directory numbers and call-park slots
Cisco Unified Communications Manager Express and Cisco Unity Express username and password synchronization
Support for IPv6 on SCCP and SIP IP phones
Enhanced security with Logical Partition Class of Restriction for Closed User Group (CUG)
Optional unlock of meet-me conference bridge
New IP Trusted Authentication List for enhanced security
Forced Authorization Code (FAC) for improved call screening and accounting
SSL VPN support for Cisco Unified IP Phones
SSL VPN support for Cisco Adaptive Security Appliance (ASA)
Fixed mobile convergence (FMC) with media flow around on SIP trunks
Support for Call Forward Unregistered
Clear directory entries for calls
Localization support for Cisco Unified IP Phone 7920 and 6900 models with SCCP
Video and camera support for Cisco Unified IP Phone 8961, 9951, and 9971 models
Increased translation rule from 15 to 100
Bulk registration support for SIP phones
Debug ephone messages enhancement
Support for Cisco Unified Wireless IP Phone 7926 wireless phone
Increased fast dial limit to 100
Support for “show voice register pool type summary”
Support for conference maximum length
Support for DX650 desktop collaboration endpoint
BACD support for voice hunt groups for SIP phones **NEW***
Voice hunt group call statistics support ***NEW***
Automatic registration of SIP phones ***NEW***
Night service support for SIP phones ***NEW***
Call-back busy subscriber and camp-on
Per-phone call-coverage rules
Call hold and retrieve
Call park: Personal and directed
Call transfer and park recall
Call park assign to extension
Call pickup directed
Call pickup local group
Call pickup explicit group
Call transfer: Consultative and blind
Call waiting
Cancel call waiting soft key
Call park and pickup on both SCCP and SIP phones
CTI with Microsoft CRM and Outlook using Cisco IOS Software Telephony Services Provider (TSP)
E911 with two emergency location numbers per zone; unlimited zones per site
Eight-party impromptu conferencing
Directory services using XML
Hunt groups: Sequential, circular, parallel (blast), and longest idle
Hunt-group dynamic log in and log out
Hunt-groups statistics: Daily and hourly
Intercom
Ad-hoc conferencing (8 parties)
Meet-me conferencing (32 parties)
Five music-on-hold (MoH) streams (internal)
Night service bell or call forwarding
Overlay extensions for enhanced call coverage
Called-name display for overlay extensions
Paging: Internal through IP phones or to external system
Per-call caller ID blocking
Secondary dial tone
Standards-based network call transfer and call forwarding using H.450
Additional system speed-dial option through XML service
Time-of-day and day-of-week call blocking
Customizable called-name display
Support of SRST fallback service phone autoregistration
Basic automatic call distributor (BACD) (three queues) with Automated Attendant and call statistics
Display of number of calls in queue on IP phone
Agent log in and log out of BACD hunt group
Integration with Cisco Unified Contact Center Express 5.0 for advanced call-center features with support for up to 50 agents, agent supervisors, call recording, silent monitoring, and reporting features
Secure Real-Time Transport Protocol (SRTP) providing media encryption for calls on the IP network
Secure voice IP phone certificate authentication and provisioning plus secure device signaling using Transport Layer Security (TLS)
Video over SIP trunk with H.264 codec support
Multilevel Precedence and Preemption (MLPP) support over PRI trunks
Enhanced MLPP with support for supplementary services, including three-way conference, call park, and call pickup
BLF monitoring of conference directory numbers and call-park slots
Cisco Unified Communications Manager Express and Cisco Unity Express username and password synchronization
Support for IPv6 on SCCP and SIP IP phones
Enhanced security with Logical Partition Class of Restriction for Closed User Group (CUG)
Optional unlock of meet-me conference bridge
New IP Trusted Authentication List for enhanced security
Forced Authorization Code (FAC) for improved call screening and accounting
SSL VPN support for Cisco Unified IP Phones
SSL VPN support for Cisco Adaptive Security Appliance (ASA)
Fixed mobile convergence (FMC) with media flow around on SIP trunks
Support for Call Forward Unregistered
Clear directory entries for calls
Localization support for Cisco Unified IP Phone 7920 and 6900 models with SCCP
Video and camera support for Cisco Unified IP Phone 8961, 9951, and 9971 models
Increased translation rule from 15 to 100
Bulk registration support for SIP phones
Debug ephone messages enhancement
Support for Cisco Unified Wireless IP Phone 7926 wireless phone
Increased fast dial limit to 100
Support for “show voice register pool type summary”
Support for conference maximum length
Support for DX650 desktop collaboration endpoint
BACD support for voice hunt groups for SIP phones **NEW***
Voice hunt group call statistics support ***NEW***
Automatic registration of SIP phones ***NEW***
Night service support for SIP phones ***NEW***
Voicemail features
Integrated voicemail and Automated-Attendant solution with Cisco Unity Express
Integration with Cisco Unity Voicemail and Cisco Unity unified messaging, or third-party voicemail integration (H.323, SIP, or dual-tone multifrequency [DTMF])
Integration with Cisco Unity Voicemail and Cisco Unity unified messaging, or third-party voicemail integration (H.323, SIP, or dual-tone multifrequency [DTMF])
International localization
Per-phone localization for up to five local languages per system, including English, Bulgarian, Chinese Mandarin and Cantonese, Croatian, Czech, Danish, Dutch, European Spanish, Finnish, French, German, Greek, Hungarian, Italian, Japanese Kanji and Katakana, Korean, Norwegian, Polish, Portuguese, Romanian, Russian, Serbian, Slovakian, Slovenian, Swedish, Turkish, Latvian, Lithuanian, Estonian, and Hebrew
Management features
Automatic assignment of extensions to phones for easy phone additions
Extension assigner, allowing for deployment of new phones using voice prompts
Single web-based GUI for moves, adds, and changes for system and integrated voicemail with three levels of GUI administration: System administrator, customer administrator, and user
Centralized network management deployments using Cisco CNS Configuration Engine
Telephony-service setup and configuration using HTML Quick Configuration Tool
Simple Network Management Protocol (SNMP) support with Cisco Unified Operations Manager or third-party management consoles
Extension assigner, allowing for deployment of new phones using voice prompts
Single web-based GUI for moves, adds, and changes for system and integrated voicemail with three levels of GUI administration: System administrator, customer administrator, and user
Centralized network management deployments using Cisco CNS Configuration Engine
Telephony-service setup and configuration using HTML Quick Configuration Tool
Simple Network Management Protocol (SNMP) support with Cisco Unified Operations Manager or third-party management consoles
APIs for development and integration
Client integration: You can connect Cisco Unified CME with Microsoft CRM and Salesforce.com using Telephony Application Programming Interface (TAPI) Windows driver based on Cisco IOS Software TSP.
Server integration: You can connect Cisco Unified CME with third-party applications using the Cisco Unified Communications Express Services Interface API. This API enables protocol interaction between Cisco Unified CME and third-party applications to perform CTI functions, including call monitoring, call control, and call provisioning with any SCCP device registered with the Cisco Unified CME. It includes two elements: embedded CTI protocol support within Cisco Unified CME and Java Function Library (referred to as the “UC Express Services SDK”) that co-resides on an external server with the third-party application. This API can support either co-located or service provider-hosted application deployment scenarios. For more information, please refer to: http://developer.cisco.com/web/ucxapi/home.
Server integration: You can connect Cisco Unified CME with third-party applications using the Cisco Unified Communications Express Services Interface API. This API enables protocol interaction between Cisco Unified CME and third-party applications to perform CTI functions, including call monitoring, call control, and call provisioning with any SCCP device registered with the Cisco Unified CME. It includes two elements: embedded CTI protocol support within Cisco Unified CME and Java Function Library (referred to as the “UC Express Services SDK”) that co-resides on an external server with the third-party application. This API can support either co-located or service provider-hosted application deployment scenarios. For more information, please refer to: http://developer.cisco.com/web/ucxapi/home.
Refurbished CISCO VG202 datasheet
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