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DSPs Transcoding and Conferencing Part 1
 
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DSPs Transcoding and Conferencing Part 1 Visit http://configureterminal.com/dsps-transcoding-and-conferencing-part-1/ In this video David Bombal demonstrates the configuration and setup of Digital Signal Processors (DSPs) on Cisco Routers
Views: 6751 David Bombal
Section 6 Media Resources
 
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CCIE Voice LoD is a pre-recorded lecture of CCIE Voice Blue print. It is the same lecture that we do during 5 days CCIE Voice Bootcamp. You will hear CCIE Voice instructor provide in dept knowdledge of CCIE Voice lab with tips and trick. It is just like takin a CCIE Voice Bootcamp from your couch. Candidate can browse through each chapter in a sequential manner or random choose the topic
Views: 1390 Faisal Khan
Investigating one-way or no-way audio: (part 1)
 
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Tracing a basic call with wireshark by www.voiceinitiate.com ------------------------------------------------------------------------------------------------ As we know, the problem of one-way audio is very popular in the word of IP telephony and when it occurs, everyone thinks its the telephony system that's got a problem. However, 90% of the time, it's the underlying network that's causing the problem. This video entry is geared towards providing a method for determining whether the phones in a a call are actually sending audio packets when users are complaining of one-way audio or no audio at all.
Views: 18758 Voiceinitiate
MTP Beginner
 
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Basic usage MTP
Views: 89 Forex Kuliah
Configuring MoH in CUCM - Configuration and Testing
 
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Configuring CUCM MoH Systemwide. Simple Steps...
Views: 1028 JBCOMP DOTNET
CUCM media resources issue
 
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No audio available. This article shows backdoor solution to the media resources issue in Cisco Unified Communication Manager which installed not on the systems with Intel CPU. You need to have a Linux Live CD.
Views: 784 Alexander Levichev
How to Configure an FXO Port to Ring VoIP Phone on CUBE and CUCM
 
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How to Configure an FXO Port to Ring VoIP Phone on CUBE and CUCM =================================================== FXO Voice Port Configuration • Unlike a T1 Card there is no configuration needed after FXO Card installation • Each FXO port will need to be configured and there are several parameters that need to be setup. • There are other configuration setting that may be needed depending on your setup and location; I have just listed the basic settings I used to setup our Ecuador office. o connection plar (opx)  This command will ring an extension from CUCM o Signal  Loopstart - Standard residential phone lines are configured as FXS loop-start. Loop-start is the default signaling on Cisco IOS FXS and FXO voice ports. In order to change it, issue the signal ground-start voice-port command. Reset the voice-port after any changes with the shutdown/no shutdown command sequence. This is the default setting. • Source - https://supportforums.cisco.com/document/15261/how-understand-and-configure-analog-fxo-or-fxs o Supervisory disconnect dualtone mid-call  supervisory disconnect - Configures supervisory disconnect signaling on the FXO port. Supervisory disconnect signaling is a power denial from the switch that lasts at least 350 ms. When this condition is detected, the system interprets this as a disconnect indication from the switch and clears the call. You should disable supervisory disconnect on the voice port if there is no supervisory disconnect available from the switch. Typically, supervisory disconnect is available when connecting to the PSTN and is enabled by default. When the connection extends out to a PBX, you should verify the documentation to ensure that supervisory disconnect is supported o Cptone [Country Code]  This command defines the detection of call-progress tones generated at the local interface. It does not affect any information passed to the remote end of a connection, and it does not define the detection of tones generated at the remote end of a connection. Use the cptone command to specify a regional analog voice interface-related default tone, ring, and cadence setting for a specified voice port. The current configured cptone setting is visible under the ‘show voice-port [port]’ command  voice class custom-cptone [WORD] • Adding to the voice-port o Voice-port 0/1/2 supervisory custom-cptone [WORD] • World Tone Database -http://www.3amsystems.com/World_Tone_Database?q=Ecuador,Audible_ring_tone o Dial-type  DTMF is set as default o trunk-group  An interface can belong to only one trunk group. Multiple interfaces can belong to the same trunk group.  Example • voice-port 0/1/3 trunk-group FXO connection plar opx 2001 caller-id enable  Dial-peer Example • dial-peer voice 1000 pots destination-pattern .T trunkgroup FXO o Ring Number Dial-peer Configuration ---------------------------------------- Because the incoming call is coming from the PBX into the FXO port, the router answers the call. We want the incoming call to ring a VoIP phone that is connected to CUCM, and because we have set connection plar opx 2001 command, the router is going to 'dial out', and we need to create a VoIP dial-peer for this. Example: dial-peer voice 5000 voip description ** POTS/FXO IN ** destination-pattern 2001 session target ipv4:10.2.21.10 (This is the IP of CUCM) dtmf-relay h245-alphanumeric no vad dtmf-relay h245-alphanumeric -This is an out-of-band DTMF relay mechanism that transports the DTMF signals using H.245, which is the media control protocol of the H.323 protocol suite. Source – (http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf-relay.html) We need to add our other dial-peers, for outbound calls through the FXO port. dial-peer voice 4000 pots trunkgroup FXO description ** TV CABLE POTS OUT LOCAL QUITO #'s ** destination-pattern [2-9]...... forward-digits all dial-peer voice 4001 pots trunkgroup FXO description ** TV CABLE POTS OUT LD/CELL #'s ** destination-pattern [01]........ forward-digits all Good Sources http://www.cisco.com/c/en/us/support/docs/voice/h323/15405-pbx-across-ip.html CUCM Configuration Checklist ------------------------------------------------- • Date/Time Group • Region • SIP Trunk • SIP Profile • Device pool for CUBE • Device pool for PHONES • Media Termination Point (MTP) • Media Resource Group (MRG) • Media Resource Group List (MRGL) • Partition • Calling Search Space (CSS) • Route Group • Route List • Route Pattern • Create H.323 Gateway
Views: 2725 W00DY1848
CUCM Basic Troubleshooting: Cisco Unified Reporting - Cluster Overview- シスコ サポート コミュニティ
 
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このビデオでは、Cisco Unified ReportingのCluster Overview レポートについて説明しています。 - Cisco Unified Reportingの概要 - Cluster Overview Reportの項目に関する説明 Cluster Overview レポートはクラスタの状況を把握するのに最適なので、SRオープン時には取得するようにしてください。 以下のサイトは、シスコのIP テレフォニー製品に関するコミュニティサイトです。最新の技術情報も掲載しています。 https://supportforums.cisco.com/community/csc-japan/voice/ucm
DTMF Issue - voice-class sip dtmf-relay force rtp-nte
 
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The issue was that when a Sprint customer called in to our UCCX IVR main auto attendant they have 2 options (1 for ENG or 2 for SP) those options work fine, no problems. But once the customer was in the sub menu the first DTMF digit pressed would not register and would only register on the 2nd press of the DTMF digit. It only happened when using a Sprint cell phone and only happened with our sub menus but not the main AA. The solution was to enter the command 'voice-class sip dtmf-relay force rtp-nte' under dial-peer voice 200 voip which is our dial-peer for CUCM. That is a hidden command and will not show up as an option in the IOS when using the help function '?' . But it will work if entered as is under a dial-peer. This Command ensures that the CUBE will always uses RFC2833 for DTMF even if it was not offered by the provider in the initial invite. Your SIP provider must support RFC2833, and lucky for us, most providers will because RFC2833 is pretty common. ***INFO*** voice-class sip dtmf-relay force rtp-nte ---------------------------------------------------------------------------------------- https://anetworkerblog.com/2011/02/06/dtmf-on-voip/ https://supportforums.cisco.com/discussion/10709181/dtmf-relay-unrecognized-command-cli Understanding DTMF --------------------------------------------------------------------------------- DTMF Relay - http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf-relay.html DTMF AND RFC 2833 / 4733 - https://andrewjprokop.wordpress.com/2013/09/27/dtmf-and-rfc-2833-4733/ Understanding DTMF negotiation and troubleshooting on SIP Trunks - https://supportforums.cisco.com/document/144711/understanding-dtmf-negotiation-and-troubleshooting-sip-trunks Configuring and debugging DTMF (RFC 2833) - https://blogs.msdn.microsoft.com/rita_z/2005/10/10/configuring-and-debugging-dtmf-rfc-2833/ ==================================================== Multiple DTMF Methods ----------------------------------------------------------------------- Multiple DTMF methods may be configured on CUBE simultaneously in order to minimize MTP requirements. If you configure more than one out-of-band DTMF method, preference goes from highest to lowest in the order of configuration. If an endpoint does not support any of the DTMF relay mechanism configured on CUBE, an MTP or transcoder is required. Cisco UCCX jtapi ports only support out of band DTMF, you can configure your cube dial-peer pointing to CUCM to use both rtp-nte and sip-kpml. SIP-KPML will be out of band and hopefully you will not need MTP. Example: Router(config)# dial-peer voice xx voip Router(config-dial-peer)# dtmf-relay rtp-nte sip-kpml Source - https://supportforums.cisco.com/discussion/12394051/dtmf-incoming-over-sip-trunk-not-working
Views: 2773 W00DY1848
GEN3 - MTP Case Study
 
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View this video to learn how a GEN3 and Managed Technology Partners managed services partnership has helped GEN3 accomplish its technology and communications goals utilizing Cisco solutions quickly and cost effectively. www.managedtech.com
Views: 283 managedtech
UCCX Labs   CUCM Device Preparation CTI Ports CTI Route Points Auto Attendant
 
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Creation of CTI Route Points, CTI Ports, CUCM Preparation
Views: 2902 CollabEngineer
DSPs Transcoding and Conferencing Part 2
 
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DSPs Transcoding and Conferencing Part 2 Visit http://configureterminal.com/dsps-tra... In this video David Bombal demonstrates the configuration and setup of Digital Signal Processors (DSPs) on Cisco Routers
Views: 4578 David Bombal
Video 6 - MTP 4.7 Basic Settings Part3.mp4
 
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Part 3 of 3 Covering the Basic Setting tab of Manage The Pip
Views: 297 BDBFX
MTP ExhaustiF
 
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ExhaustiF is a tool for grey box testing based on the technique SWIFI (SW Fault Injection) used to improve reliability and availability in critical systems. The tool must be used during system integration and system validation phases. ExhaustiF tool is for those companies that develop critical software intensive systems for Aerospace, Defence, Military and Railway Industries.
Views: 31 MTP dba
SP-DAY2-4-MPLS-TE
 
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Views: 3495 ralph loren
7. - Conference Bridge  Creation & Management
 
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How to configure the inbound conference bridge functionality on the UNIFIED TELECOM hosted platfrom
Views: 470 UnifiedTelecom
Denny Trevett - Cisco Systems with MTP
 
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Denny Trevett, Cisco Systems, what are businesses doing in technology today: Unified Communications, Virtualization and Web 2.0
Views: 275 managedtech
CCIE Collaboration Lab 3 Demo Section 3 4a Hardware Conference Resources
 
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Cisco Collaboration Training - 100% Focus on UC Collaborations
Views: 430 VoiceBootcamp Inc
Video 5 - MTP 4 7 BasicSettings Part2
 
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Part 2 of 3 Covering the Basic Setting tab of Manage The Pip
Views: 304 BDBFX
Trace analysis of DSP
 
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Views: 407 imec
MWC2012 Polycom VVX1500 using Ericsson MRFP and early media
 
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MWC2012 Polycom VVX1500 using Ericsson MRFP and early media
Views: 500 Amir Levy