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PVDM-12

PVDM-12

CISCO Refurbished

Manufacturer
CISCO
Condition
Refurbished
Availability
In Stock
List Price/RRP
US$1,795.00
Price
US$20.00 Enquire

Description

The PVDM-12 is a product from CISCO that enhances the voice and video capabilities of their networking devices, specifically their routers. It is a type of Digital Signal Processor (DSP) module that allows for improved voice and video quality, as well as increased capacity for multiple calls and video streams. It essentially acts as an accelerator for voice and video processes on the network, reducing latency and ensuring smooth communication. The module has 12 channels, which means it can handle up to 12 simultaneous voice or video calls. The PVDM-12 is compatible with various CISCO routers and can be easily installed using a simple plug-and-play method. It is a highly recommended upgrade for businesses or organizations that rely heavily on voice and video communication, such as those in the healthcare, education, and finance industries. With the PVDM-12, users can expect improved call quality, more efficient use of network bandwidth, and better overall network performance.

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about PVDM-12

We are Refurbished Network Equipment specialists and we may have PVDM-12 available to us if it’s not in stock. Please complete the form and we'll get back to you as soon as we can. We have a vast amount of experience and knowledge with all legacy CISCO components and will be happy to provide you with expert after-sales advice, should you need it.

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Part Code
PVDM-12
Description
12-Channel Packet Voice/Fax DSP Module
Weight
0.01kg
Manufacturer
CISCO
End of Hardware Support
Yes
End of Sale
Yes (25 September 2006)
End of Life
Yes (27 March 2006)
Warranty
2 year warranty icon

Digital T1/E1 Packet Voice Trunk Network Module Feature Summary

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Digital T1/E1 Packet Voice Trunk Network Module Feature Summary

Feature
Benefits
Scalable from 6 to 60 Voice Channels
Single Network Module scales using from one to five 12-channel packet voice DSP modules (PVDM-12) upgrade SIMMs to support from 6 to 60 voice channels.
Platform Voice Scalability to 300 Voice Channels
Enables Cisco 2600, 2600XM, 2691, 2800, 3600, 3700, 3800 series modular access routers to scale from 6 to 300 voice channels in a single multiservice router solution.
Network Module
Enterprises and service providers can use a single box to support data and telephony services by sharing this Network Module with other interfaces in the Cisco 2600, 2600XM, 2691, 2800, 3600, 3700, 3800 series modular access routers.
Standards-Based PCM Encoding
Standards-based ITU-T G.711 PCM encoding provides 64 kbps analog to digital conversion using u-law or A-law.
Standards-Based Compression Algorithm Support
Users can choose to either transmit voice across their networks as uncompressed PCM (G.711, u-law and A-law) or compressed from 5.3 kbps to 32 kbps using standards-based compression algorithms (G.729, G.729a/b, G.723.1, G.726, G.728).
Fax Support
Transmit Group III fax over any voice channel without sacrificing voice processing resources regardless of compression type.
Voice Over IP
Transmit data, voice, and video across a single frame relay, ATM, ISDN, channelized, or multilink point-to-point protocol (MLPPP) network.
Voice Over Frame Relay
Leverage existing or new frame relay network by transporting voice directly over this network using standards based transport methods (FRF.11) combined with standards based fragmentation for data (FRF.12). VoIP can also be transmitted over Frame Relay.
Voice Over ATM
Transport voice directly over ATM networks using AAL2 and AAL 5 encapsulation. Leverages existing ATM networks as a direct transport method for voice. VoIP can also be transported over ATM (VoATM requires ATM network modules such as IMA or OC-3).
Connection Trunk
Creates a tie-line replacement structure while only consuming bandwidth during a call (digital-to-digital, digital-to-analog, or analog-to-analog capabilities).
Toll Bypass
Reduce or eliminate toll charges assessed by long distance and local carriers by transporting voice and fax traffic across the enterprise intranet, LAN, metropolitan-area network (MAN), or WAN.
Integrated Data WAN Support
Connect one or two T1 or E1 interfaces on this network module as a WAN interface either as one or multiple groups of DS0's or as an entire T1 or E1 frame while still providing packet voice support for connections to PBXs and the PSTN.
LVBO (Local Voice Busy-Out)
Automatically busy out any desired voice trunk line (or individual DS0s) to a PBX or PSTN when a direct WAN or LAN connection to the router is down. Also, busy out a far end trunk connection when configured for Connection Trunk.
Call Admission Control Using RTR
Uses Response Time Reporter (RTR) to determine latency, delay and jitter and provide real-time ICPIF calculations before establishing a call across an IP infrastructure. RTR packets emulate voice packets receiving the same priority as voice throughout the entire network. A superior method to data and ping packets for determining congestion levels.
Off-Premise Extension (OPX)
Extends the capability of legacy PBX to off-premise phones.
Voice Activity Detection (VAD)
Consume bandwidth during a call only when there is voice traffic to send (silence suppression)
Comfort Noise Generation
While using VAD, the DSP at the destination emulates background noise from the source side, preventing the perception that a call is disconnected.
Circuit Switched Leased Line Replacement
Businesses incur significant recurring monthly costs for leased lines purely for the interconnection of telecom PBXs and switches. This product gives these enterprises the ability to remove these costly rigid-bandwidth leased lines and replace them with flexible bandwidth lines, which will be used to carry data, voice, and video. The ability to support proprietary PBX signaling types exists by using connection trunking and transparent CCS.
Private Line Automatic Ringdown (PLAR)
Provides a direct connection to another digital or analog voice port by lifting a telephone handset on one end.
H.323 Version 3 and 4 Support
Uses industry-standard signaling protocols for call setup between gateways, gatekeepers, and H.323 end points (such as Intel Internet Phone or Microsoft Netmeeting).
DTMF Relay
Carries DTMF tones/information out-of-band for clearer transmission and detection.
Authentication, Authorization, and Accounting (AAA)
Supports debit and credit card (prepaid and post-paid calling card) applications.
Interactive Voice Response Support (IVR)
Utilizes IVR to provide automated-attendant, voice-mail support, and call routing based on desired service.
Open Settlement Protocol Support (OSP)
Provides the ability to settle account billing between service providers who are sharing resources to expand geographical coverage using third-party tools and standards-based OSP.
Any Call to Any Call with End-to-End Interoperability
Interoperates with Cisco IP phones, analog phones, fax machine connections, and PBX connections to and from any other Cisco voice enabled product.
Gateway for Legacy PBXs, Phones, Fax Machines, and Key Communication Systems to PSTN
Enables a connection for incoming and outgoing calls to and from the PSTN originating from and destined for legacy PBXs, phones, fax machines, and key communication systems connected to a data, voice, and video infrastructure.
Integrated Add/Drop Multiplexer
Performs add/drop multiplexing for voice within a dual-port voice Network Module. Eliminates the requirement, maintenance, support, and expense found when using an external add/drop multiplexer.
AVVID Interoperable
Interoperable within Cisco's AVVID architecture.
Call Control Signaling
Supports H.323 v3/v4, MGCP, and SIP call control protocols.
Gateway for IP phones to PSTN or Classic PBXs
Enables a connection for incoming and outgoing calls to and from the PSTN or classical PBXs using Cisco IP phones.

Digital T1/E1 Packet Voice Trunk Network Module Technical Specifications

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Digital T1/E1 Packet Voice Trunk Network Module Technical Specifications

Specification
Standard
Quality of Service (QoS) Standards
WFQ, WRED, CRTP, LLQ, Diffserv, RSVP, and other IOS capabilities
Signaling Standards
ITU-T: H.323 v3 and v4
MGCP, SIP
T1 CAS
E&M Wink Start, Immediate Start, Delay-Dial
FXS Loop-Start, Ground-Start
FXO Loop-Start, Ground-Start
Fax
T.38 & T.37
Clock Support
Pull-in range 64 PPM, Pass-through 32 PPM
Safety Standards
UL 1950 3rd. edition
CSA 950, 1995 version
IEC 950
EN 60950
AS/NZS: 3260 with amendment 134
Maximum Simultaneous Call Setup
60 Calls per Network Module
Interface Type
RJ48 Connector
Telco Standards
AT&T Accunet (62411)
ATT 54016
Line Bit Rate
T1, 1.544Mbps
E1 2.048Mbps
Line Code
AMI, B8ZS (T1)
HDB3 (E1)
AMI Ones Density
Enforced for N x 56kbps channels
Framing Format
D4 (SF) and ESF
Output Level (LBO)
0, -7.5, or -15 dB
Input Level
_ down to -24 dB0
Line Frequency
1.544 Mbps 75 bps/32PPM
2.048 Mbps 75 bps/32PPM
DTE/DCE Interface (VIC Mode)
G.704/structured
Diagnostic Loopback Support
ANSI T1.403 Annex B/V.54 loopup/down code recognition, network loopback, and user initiated loopbacks, network payload loopback, local DTE loopback, remote line (codes: V.541, loop up, and loop down)
Alarm Detection
Alarm indication signal (AIS), remote alarm, far-end block error (FEBE), out of frame (OOF), cyclic redundancy check (CRC) multiframe OOF, signaling multiframe OOF, frame errors, CRC errors, Loss of network signal (red alarm), loss of network frame, receive (blue alarm) (AIS) from network, receive (yellow) from network Performance Reports/Error Counters CRC, errored seconds, burst errored seconds, severely errored seconds, Ft and Fs framing errors for SF framing, FPS framing errors for ESF framing, 24-hour history stored in 15-minute increments
LED Indicators
Data carrier detect (CD)
Loopback (LP)
Alarm (AL)
Voice DSP processing status
DSU/CSU
Selectable DSX-1 cable length in increments from 0 to 655 feet in DSU mode
Selectable DS1 CSU line build-out: 0, -7.5, -15, and -22.5 dB
Selectable DS1 CSU receiver gain: 26 or 36 dB
Physical Interface Standards
T1 ANSI, ATT T1.1, ANSI T1.403
Environmental
Operating temperature: 0 to 40º C (32 to 104° F)
Storage temperature: -25 to º C (-13 to 158° F)
Relative humidity: 5 to 85% noncondensing operating; 5 to 95% noncondensing, nonoperating
MTBF
NM-HDV-1T1-24E; 381,087 to 467,253 hours
NM-HDV-2T1-48; 366,394 to 445,354 hours

Digital T1/E1 Packet Voice Trunk Network Module Management

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Digital T1/E1 Packet Voice Trunk Network Module Management

Telnet/Console
Remote and local configuration, monitoring, and troubleshooting from Cisco IOS CLI
SNMP
Router and DSU/CSU managed by single SNMP agent
Router/DSU/CSU appear as single network entity to user standard MIB (MIB II)
Cisco integrated DSU/CSU MIB
RFC 1406 T1 MIB, including alarm detection and reporting
SNMP Traps
Generated in response to alarms

Digital T1/E1 Packet Voice Trunk Network Module T1 Interface Country Approval

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Digital T1/E1 Packet Voice Trunk Network Module T1 Interface Country Approval

Country
Specification
US
US (UL 1950, T1)
FCC Part 68
FCC Part 15 Class B, T1
Canada
CS-03
CSA 950, T1
CSA C108.8 Class A, T1
Japan
VCCI class 2
VCCI:V-3/97.04, T1
JATE green book
IEC950

Digital T1/E1 Packet Voice Trunk Network Module E1 Interface Country Approval

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Digital T1/E1 Packet Voice Trunk Network Module E1 Interface Country Approval

Country
Specification
Austria (CE) Belgium (CE Denmark (CE) Finland (CE) France (CE) Germany (CE) Gibraltar (accepts CE) Greece (CE) Ireland (CE) Italy (CE) Liechtenstein (accepts CE) Luxembourg (accepts CE) Malta (accepts CE) Monaco (accepts CE) Netherlands (CE) Norway (CE) Portugal (CE) Spain (CE) Sweden (CE) Switzerland (CE) U.K. (CE)
EMC EN55022/EN50082/EN61000
Safety EN60950
Telecom CTR13/CTR12
New Zealand
EMC AS/NZS 3548
Safety AS/NZS 3260
Telecom TNA 117
China
EMC CISPR22
Safety EN60950
Singapore
EMC EN55022/CISPR22
Safety EN60950
Telecom DLCN1/DLCN2
Poland
EMC EN55022/EN50082/EN61000
Safety EN60950
Telecom CTR13/CTR12
Australia
EMC AS/NZS 3548
Safety AS/NZS 3260
Telecom TS016

Why Buy Refurbished?

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Product Details

We provide refurbished Cisco kit, finished to exceptional standards: rigorously tested, immaculately packed and backed by the best customer service you'll find anywhere.

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Committed to Cisco

Go Communications specialises in Refurbished Cisco Systems equipment. We can source virtually any Cisco part. Through our SAMEasNEW refurbished programme, we can even provide high quality, pre-owned parts that Cisco no longer makes - reducing your need to upgrade end-of-life equipment.

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Committed to quality

Go Communications is committed to the highest levels of reliability and quality. Our state-of-the-art Cisco Certified Internetwork Expert (CCIE) development facility 'soak tests' every device and component we sell: only if it meets our exceptionally high standards can it be accepted onto our market-leading two-year warranty programme.

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Committed to service

We believe in honouring our commitments, providing outstanding quality and delivering fantastic customer service at prices which are very hard to beat. As a result we have built a reputation as a trusted, reliable and knowledgeable source of high quality refurbished Cisco components.

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Experience

Everyone at Go Communications has been involved with Cisco Systems products for more than fifteen years. That's not just those in our sales and engineering teams, but also our purchasing, logistics and even finance departments.

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Fast Delivery

We believe we are one of the UK's fastest and most efficient Cisco delivery services. For instance, 'in stock' items can be delivered within the M25 corridor within two hours, or to Scotland and Wales within eight hours.

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Two Year Warranty

We offer an industry-leading two year warranty* on all of our refurbished products. This includes overnight advanced replacement and ongoing technical support from our CCIE and CCNA certified technicians. We aim to replace all equipment under warranty within 24 hours. If the item isn't in stock, please allow up to 72 hours. If a replacement isn't available in time, we may offer a temporary solution free of charge. * Retail end-user clients only.

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Please follow the links for more details on our Terms of Purchase and Terms of Sale. For more information about our products, please call 01279 408 777 or email us at: sales@gocomsys.com.

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