_title_: Senior VOIP Engineer Location: City: Jersey City, State: New Jersey, Zip: 7097 Max Rate: 75 $/hr - 80 $/hr Apply Here:
http://www.staffitnow.com/StaffIT/JobServlet?internal=Y&ref=656508 De_script_ion: (Visit the site for detailed de_script_ion) SR Voice Consultant The customer will provide a template for the configuration work to be performed. The successful candidate must be able to complete the following tasks. On the Cisco Call Manager: Partitions Calling Search Spaces Device Pool SRST Reference (Use loopback 5 IP Address for the SRST Router) Route Group Route List Route Patterns Conference Bridge Media Resource Group Media Resource Group List Voicemail Profile Gateway Configuration On the Voice Gateway Router: network-clock-participate and network-clock-select command options for circuit clocking dspfarm and dsp services dspfarm command options for Hardware DSP resources T1/E1 Controllers for PRIs, QSIG Connections to IPC IP PBX. Build appropriate dial-peers for Hoots and ARDs for connectivity to the End-End Hoot/ARD routers _base_d in New York. (Please refer to the detailed process of building Hoots/ARDs FXS Ports for Analog Stations ISDN Interface for PRI Backhaul to CCM CCM-Manager fallback-MGCP Command option for MGCP fallback on Voice Gateway MGCP Configuration H.323 Dial-peers for SRST Call routing SRST Configuration Place a telephone call to the Carrier Technician for the actual PRI Cut-over. Ensure that all PRIs are turned up properly. _layer_s 1, 2 and 3 must be UP and active. Please use show isdn stat command on the voice gateway to confirm this status. Ensure that both inbound and outbound calls are working by thoroughly testing with the test DIDs provided by the Carrier. Configure the QSIG Pipes on the CCM Gateway Admin Page to the IPC IP PBX. Work with IPC Technician to ensure that proper connectivity is established between the IPC IP PBX and the Call Manager. Build appropriate dial-peers on the Voice gateway to route calls from the IPC IP PBX to the CallManager (for MGCP and H.323 Call processing) Ensure that dial-tone is provided by the CallManager to the IPC IP PBX and appropriate call digits are being sent from the IPC IP PBX to the CallManager. Build the Site Call Processing System Integration with Cisco Unity Voicemail System. After the Telecoms Analysts build voicemail boxes for individual users for the site, ensure that appropriate Translation Patterns and Call Search Spaces are configured on the CallManager to route calls between 5 digits from the CallManager to 7 digits on the Cisco Unity Voicemail System Build Call Manager configuration for creating voicemail profile Build CallManager configuration for MWI Configure the appropriate Translation Patterns required for the OnNet Calls and also configure the Gatekeeper to recognize these translation patterns for the Site OnNet Call routing. With the Assistance of the Site Telecom Analyst, carry out a comprehensive VoIP test for all call features as specified in the Cut-over Spreadsheet for the Site. This will also include Hoots, ARDs and SRST Test. Complete the Site VoIP Implementation Documentation appropriately and upload to SharePoint. CCNP + 5 years voice implementation experience. CCVP or CIE Voice a plus. Call Manager 4.x configuration (hands on), PRI implementation (work with carriers to test and turn up pri s to the Cisco equipment), CSU/DSU configuration. End user phone usage training Create as built documentation (configurations and Visio diagrams) VoIP Gateway router configuration Q-Sig configuration MWI configuration CCNP or CCVP + 5 years voice hands on. Call Manager 4.x configuration. Carrier ckt troubleshooting. View mail configuration a plus. Q-Sig integration MS Share-point experience a plus
