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UCCX HIGH AVAILABILTY

December 18, 2012 Leave a comment

UCCX HIGH AVAILABILTY

 Step 1  Power ON the Secondary Server and start the services.

Step 2  Upload HA license to Publisher.

System > License Information > Add License.

Step 3  Add new server to the cluster.

System > Server>Add New.

Step 4  Select Network   Deployment Type – WAN/LAN

System > System Parameters.

Step 5  Select Failover Server.

Cisco Unified CCX Serviceability>Tools>Service parameters>ALPHAUCCX>Cisco AMC Service>Failover collector>

Step 6  Check the DB replication status for the data stores.

             Cisco Unified CCX Serviceability>Tools > Data store  Control Center > Replication Servers  

Step 7  Create CTI ports for Secondary Server.

Step 8  Reboot the Servers.

Categories: CCIE VOICE, RealTime DB, Troubleshoot Tags:

Cisco Nexus 1000V Webinar Training Video’s

October 24, 2012 1 comment

Nexus 1000V Webinar Training Videos
Nexus 1000V Family Overview and Update
Virtual Services (VSG, vWAAS, vNAM)
Virtual Security Gateway Introduction
Journey to the Cloud w/ N1KV: vCloud Director & Long Distance vMotion
Secure VDI with Nexus1000V & VSG
Nexus 1000V v1.4 New Features and Installation/Upgrade Overview
Nexus1010 Overview & Best Practices
Virtual Security Gateway Technical Overview
Nexus 1000V Key Features Overview
Nexus 1000V Troubleshooting
Nexus 1000V Web Site
Nexus 1000V Webinars
Nexus 1000V and VMware vCloud Director Whitepaper
Nexus 1000V and VMware View Cisco Validated Design
Cisco Virtual Experience Infrastructure Configuration Guide (including Nexus 1000V)
Virtualize DMZ Whitepaper
Nexus 1000V Community
Cisco Nexus 1000V Virtual Switch Software Demo Video
VMware and Cisco Accelerate Virtual Machine Deployments
Cisco Nexus 1010 Virtual Services Appliance
Cisco Nexus 1000V Network Analysis Module Virtual Service Blade 4.2.1N
Cisco Nexus 1000V Series Switches
Cisco Nexus 1000V Series Switches Video Data Sheet
Cisco Virtual Network Management Center
Cisco Virtual Security Gateway for Cisco Nexus 1000V Series Switches
Cisco Virtualization Experience Infrastructure (VXI) Configuration Guide
Cisco Nexus 1000V MIB Quick Reference
Cisco Nexus 1000V Series Switches Deployment Guide Version 2
Best Practices in Deploying Cisco Nexus 1000V Series Switches on Cisco UCS B and C Series Cisco UCS Manager Servers
Cisco Nexus 1000V Command Reference, Release 4.2(1)SV1(4)
Cisco Nexus 1000V Installation Screencasts
The following screencasts are available to assist you in the installation of Release 4.2(1) SV1(4) of the Cisco Nexus 1000V.
Understanding Cisco Nexus 1000V and VMware Software Version Compatibility
Video This screencast introduces and explains the use of the Host Software Version Compatibility table in the Cisco Nexus 1000V and VMware Compatibility Informationdocument provided for each release. Given the VMware host ESX or ESXi software version, the compatibility table shows the appropriate Virtual Ethernet Module (VEM) installation files and the minimum required software versions of the VMware vCenter Server, the vCenter Update Manager, and the vSphere CLI. The screencast also shows how to determine the software version of various components and how to download update software.
Installing a Redundant Pair of VSMs from an OVA file Using the Installer Application
Video This screencast shows how to install a redundant pair of VSMs in an HA configuration from an OVA file using the Installer Application.
Installing the VEM using the VUM [two parts]
VideoVideo Part 1 of the screencast shows you how to install the VEM on an ESX host using the VUM.Part 2 of the screencast shows you how to troubleshoot and verify the installation.
Installing the VEM using the vSphere CLI
Video This screencast shows how to install the VEM on an ESX host using the VMware vSphere CLI.
Cisco Nexus 1000V Upgrade Screencasts
The following screencasts are available to assist you in upgrading the Cisco Nexus 1000V to Release 4.2(1) SV1(4). We recommend that you view these screencasts and perform the procedures in the order shown.
Understanding Cisco Nexus 1000V and VMware Software Version Compatibility
Video This screencast introduces and explains the use of the Host Software Version Compatibility table in the Cisco Nexus 1000V and VMware Compatibility Informationdocument provided for each release. Given the VMware host ESX or ESXi software version, the compatibility table shows the appropriate Virtual Ethernet Module (VEM) installation files and the minimum required software versions of the VMware vCenter Server, the vCenter Update Manager, and the vSphere CLI. The screencast also shows how to determine the software version of various components and how to download update software.
Understanding the Upgrade Process for the Cisco Nexus 1000V Release 4.2(1) SV1(4)
Video This screencast provides an overview of the steps required to update your VMware infrastructure for a non-disruptive upgrade, and to update your VEMs and VSMs to Cisco Nexus 1000V Release 4.2(1) SV1(4).
Upgrading the VMware vCenter Update Manager to Release 4.0 Update 1 Patch 2
Video This screencast shows how to upgrade the VMware vCenter Update Manager (VUM) to the minimum required version for the upgrade to Cisco Nexus 1000V Release 4.2(1) SV1(4).
Upgrading the Cisco Nexus 1000V from 4.0(4) SV1(3, 3a, or 3b) to 4.2(1) SV1(4) [three parts]
This three-part video shows you how to upgrade your Virtual Ethernet Modules (VEMs) and Virtual Supervisor Modules (VSMs) to Cisco Nexus 1000V Release 4.2(1)SV1(4).
Video Part 1 of the screencast describes the process, lists the prerequisites, and shows how to download the required software.
Video Part 2 of the screencast shows you how to upgrade the VEMs.
Video Part 3 of the screencast shows you how to upgrade the Virtual Supervisor Modules (VSMs).
Upgrading the Cisco Nexus 1000V from VMware Release 4.0 to 4.1 [three parts]
This screencast shows how to upgrade your vCenter Server, vCenter Update Manager, and vSphere CLI from VMware Release 4.0 to Release 4.1.
Video Part 1 describes how to check software compatibility and how to upgrade the vCenter Server.
Video Part 2 describes how to upgrade the VUM.
Video Part 3 describes how to upgrade the hosts.

CCNP Voice free Training

October 24, 2012 3 comments

All About Extension mobility

October 11, 2012 3 comments

Introduction

This document describes the common problems in Extension Mobility.

Error :- Host not found

Resolution:-

 

  • Check that the Cisco Tomcat service is running by choosing Cisco Unified Serviceability > Tools > Control Center—Network Services

 

  •                          If you have changed the ip address on service URL then click on “Update subscriptions” (Device > Device Settings > Phone Services >IP Phone Services Configuration).and resubscribe each phone to which the wrong service was subscribed.

 

 

 

Error:- You can’t see the EM feature after hitting the services button

 

Resolution:-

 

-Verify that you have configured the Extension Mobility service

-Verify the service URL is correct

– Start/Restart the EM services on each node you are running.

 

 

 

Error:- You can’t log in/out of the EM feature but you can see it after pressing the services button

Resolution:-

 

This error comes when you haven’t enabled the extension mobility , subscribed the phones/device profiles to the service as needed and haven’t associated user to a device profile.

 

 

 

 

 

Error:- To set up speed dials and other services from your phone, please goto https://x.x.x.x:8443/ccmuser/showHome.do

Resolution:-

The above error comes when you haven’t subscribed the phone or device profile to the EM profile. Once this is done you should be able to see the EM profile and log in correctly.

Error:-After a user logs out and the phone reverts to the default device profile, the user finds that the phone services are no longer available.

Resolution:-

1. Check the Enterprise Parameters to make sure that the Synchronization Between Auto Device Profile and Phone Configuration is set to True.

2. Subscribe the phone to the Cisco Extension Mobility service.

 

 

Error:-After performing a login or logout, the user finds that the phone resets instead of restarting.

Resolution:-

  • Locale change may provide the basis for reset.

 

  • If the User Locale that is associated with the login user or profile is not the same as the locale or device, after a successful login, the phone will perform a restart that is followed by a reset. This occurs because the phone configuration file is being rebuilt.

Error[201]-Authentication error

Resolution:-

The user should check that the correct UserID and PIN were entered; the user should check with the system administrator that the UserID and PIN are correct.

Error [22]-Dev.logon disabled

Resolution:-

Make sure that you have chosen “Enable Extension Mobility” check box on the Phone Configuration window.

Error [205]-User Profile Absent

Resolution:-

Make sure that you have associated a Device Profile to the user.

Error [208]-EMService Conn. error

Resolution:-

Verify that the Cisco Extension Mobility service is running by choosing Cisco Unified Serviceability > Tools > Control Center—Feature Services

Error [25]-User logged in elsewhere

Resolution:-

Check whether the user is logged in to another phone. If multiple logins need to be allowed, ensure the Multiple Login Behavior service parameter is set to Multiple Logins Allowed

Error:- Http Error [503]

Resolution:-

 

If you get this error when Services button is pressed, check that the Cisco Communications Manager Cisco IP Phone Services service.

  • If you get this error when you select Extension Mobility service, check that the Cisco Extension Mobility Application service is running by choosing Cisco Unified Serviceability > Tools > Control Center—Network Services.

 

 

Error:- [202]-Blank userid or pin

Resolution:-

Enter a valid userid and PIN.

Error:- [26]- Busy, please try again

Resolution:-

  • Check whether the number of concurrent login/logout requests is greater than the Maximum Concurrent requests service parameter. If so, lower the number of concurrent requests.

 

  • To verify the number of concurrent login/logout requests, use Cisco Unified Communications Manager Cisco Unified Real-Time Monitoring Tool to view the Requests In Progress counter in the Extension Mobility object.

Error:-[6]-Database Error

Resolution:-

 

  • Check whether a large number of requests exists
  • If large number of requests exists, the Requests In Progress counter in the Extension Mobility object counter specifies a high value. If the requests are rejected due to large number of concurrent requests, the Requests Throttled counter also specifies a high value.

Error:- [207]-Device Name Empty

OR

Error:- XML Error [4] Parse Error

Resolution:-

Check that the URL that is configured for Cisco Extension Mobility is correct and there should be no space in between.

Error:- 8945 phone does not show EM service

Resolution:-   Set service provisioning to default or internal. Refer Bug CSCtx70127

Categories: CCIE VOICE, Troubleshoot

Distributed Voice and VXML Gateway Design with CVP SIP Deployments

September 9, 2012 3 comments

Introduction

This document discusses solutions to distributed VXML and Voice GW deployments when Ingress-GW and VXML-GW are NOT residing on same router in SIP call flow scenarios.

 

When the VoiceXML and voice gateway functions reside at the same branch location but on separate devices, an IPCC design engineer should make sure that the VRU leg is sent to the local VoiceXML GW. There are two ways to ensure that the calls are handled within the branch and not sent across the WAN to a different VoiceXML gateway.

Two possible solutions

  • Configure Unified ICM with multiple Customers instances, one per location
    • Since this is not a scalable solution and also cannot be used if VRU is Type10 so we will not discuss it here.
  • Configure Unified CVP with setTransferLabel or SigDigits feature
    • If a customer has VoiceXML Gateway and Voice Gateway configured on the same router, then you should use CVP Call Server’s “setTransferLabel” mechanism which applies only to co-resident VoiceXML and Voice Gateway configurations.
    • If a customer has VoiceXML Gateway and Voice Gateway configured on different routers, then you should use CVP Call Server’s “SigDigits” mechanishm which applies to distributed VoiceXML and Voice Gateway configurations.

Unified CVP SigDigits Feature

The SigDigits feature in Unified CVP allows you to use the dial plan on the SIP Proxy to route calls to the correct site. When the call arrives at an ingress gateway, the gateway will prepend digits before sending the call to Unified CVP. Those prepended digits are unique to that site from a dial-plan perspective. When the call arrives at Unified CVP, Unified CVP will strip the prepended digits and store them in memory, resulting in the original DID on which the call arrived. Unified CVP then notifies Unified ICM of the call arrival using the original DID, which matches a Dialed Number in Unified ICM.

 

Example

 

Call arrives at the Toronto voice gateway (TOR-V-GW) with a DID of 416-431-1000. The Toronto Voice Gateway will add 821 as a site code and will send it to central SIP Proxy server in Chicago. The Chicago SIP Proxy will send it to Chicago CVP server. So Chicago CVP Server will receive 821.416-431-1000. On the Chicago CVP server we have “SigDigits=3″ configured. Chicago CVP server will strip 821 and will send 416-431-1000 as a dialed number to ICM

 

Unified ICM Returns a Label

When Unified ICM returns a label to Unified CVP in order to transfer the call to a VoiceXML gateway for IVR treatment or to transfer the call to an agent phone, Unified CVP will prepend the digits that it stored in memory before initiating the transfer. The dial plan in the SIP Proxy must be configured with the prepended digits in such a way to ensure that calls with a certain prepended digit string are sent to specific VoiceXML gateways.

 

Example

 

ICM returns 416-431-2000 as label to queue the call at the edge Toronto VoiceXML gateway (TOR-VXML-GW). ICM will add 821 and will send the call to SIP Proxy server. Chicago SIP Proxy server will receive 821.416-431-2000. Based on the 821 site code, it will send call to TOR-VXML-GW for queuing.

 

How to turn on SipDigits Feature

To turn on this feature and specify how many significant digits should be stripped is to change the sip.SigDigits = X field in sip.properties file in all CVP Call Servers. Where X is the number of digits to be stripped.

 

SIP RingTone and Error Services

There is one caveat that SigDigits feature does not cover the 91919191 (RingTone) and 92929292 (Error) DNs for those services. What this means is that once the agent becomes available, CVP needs to pull the call that is queued at the edge on the VXML-GW and connect the caller to the agent. During this transition, CVP needs to play a ringback tone to the PSTN caller. CVP does that by initiating a SIP call (CVP sends a SIP invite) towards the VXML-GW by dialing 91919191 to play a rintone back.

 

VXML-GW matches this 91919191 on an incoming SIP VoIP dial-peer and sends ringback.wav file to the ingress Voice GW. This file is being stored in the VXML-GW and is benign sent to the ingress GW. So there will be g711 traffic will flow for some duration.

Categories: CCIE VOICE, RealTime DB, Troubleshoot Tags:

H.323 Call Flow for CVP Comprehensive Deployment Model

September 9, 2012 Leave a comment

Introduction

This document discusses very high level and brief over view of H.323 Call Flow in CVP Comprehensive Deployment Model.

The PSTN call could arrive using a traditional T1/E1 PRI trunk or using some IP based trunk potentially a SIP trunk. The PSTN call will be terminated on a Cisco voice gateway in case of T1/E1 PRI trunk for example. And from the voice gateway it could be sent to a H.323 gatekeeper or directly to the CVP depending on the deployment.

 

  PSTN----T1/E1-----VGW----GK----->CVP

OR

 

  PSTN----T1/E1-----VGW----------->CVP

In case of SIP trunk, it will be terminated on a Cisco Unified Border Element (also called Session Border Controller). From here it could be sent to a Cisco Unified Presence Server (CUPS) SIP Proxy Server or directly to the CVP depending on the deployment.

 

  PSTN----SIP------CUBE----CUPS-SIPProxy----->CVP

OR

 

  PSTN----SIP------CUBE---------------------->CVP

 

 

Call Flow

1. Public switched telephone network (PSTN) call arrives at the ingress voice gateway (VGW).

2. Ingress VGW delivers the call to CVP (with H.323).

3. CVP informs ICM of the new call. This leg is the switch leg of the call. ICM routing script instructs CVP to perform queuing or self-service application (using CVP studio).

 

  CVP-----Switch-Leg----->CUICM

4. CVP transfers the call to the Voice Extensible Markup Language (VoiceXML) gateway for VRU leg operation to perform prompt/queuing or self-service. VoiceXML gateway fetches VoiceXML pages from the CVP Call Server and/or CVP VoiceXML Server.

 

  CVP-----VRU-Leg-------->VXML-GW

5. After the self-service session or queuing, eventually ICM sends an agent label (directory number) to CVP to transfer the call to the agent.

 

  CUICM-----agent-label------>CVP

6. CVP tears down the VRU leg towards the VoiceXML gateway and transfers the call to the agent through IP switching by performing the call setup (H.323) to the destination IP device (egress VGW or Communications Manager, etc.) and redirecting the switch leg media stream to the agent destination. CVP remains in the call signaling path for subsequent transfers as needed and the media flows from the ingress VGW to the destination IP endpoint.

 

Categories: CCIE VOICE, RealTime DB, Troubleshoot Tags:

SIP Call Flow with CVP Comprehensive Model

September 9, 2012 1 comment

Introduction

This documents aims to provide detailed SIP CVP comprehensive Call Flow with the debugs captured from the CVP logs and IOS/VXML Gateways

Network Setup

The setup is very simple to demonstrate the SIP call flow.

  • A call comes in from PSTN Phone and goes to the ingress gateway
  • Ingress gateway is also acting as VXML Gateway for this setup
  • Ingress gateway sends the call to CUP SIP Proxy
  • CUP SIP Proxy sends the call to the CVP Call Server
  • CVP Call Server will send the call to ICM
  • Agents are connected to the CUCM

Following script is being used (Notice that Microapps are being used and no Call Studio script is being used in this setup)

ICM Script Flow

  • Call comes in and hit Start
  • Some variables are being set (HTTP Media Server IP Address, Directory Name for the stored media files, en-us locale, input type being set it digits etc.)
  • Then call is being sent to VRU which VXML-GW
  • VXML instructions are being sent to the VXML-GW
  • Save the digits into a variable
  • Play those digits back to the caller
  • Set LAA (Longest Available Agent) agent selection algorithm
  • Wait in the queue for agent to become available
  • During this time play a hold music to the caller in the loop

In the following section the details will be discussed step by step

(1) Call Comes in from the PSTN

PSTNph---PSTN----Ingress-gw----->CUPS

Call matches an incoming pots dial-peer for the destination number. It is important to configure cvp-survivability service under the POTS dial-peer. If this service is not configured on the incoming pots dial-peer, the ingress gateway will not be able to communicate with the CVP Call Server and might receive “SIP/2.0 503 Service Unavailable” message from the CVP Call Server.

dial-peer voice 1 pots
service cvp-survivability
incominca-called-number 5417641400
direct-inward-dial

In the event of some critical CVP application failure, survivability.tcl script will attempt to recover and smoothly disconnect the caller after playing critical_error.wav file from Ingress-GW’s flash. It is very important to have this script because this script also plays pleasewait.wav and holdmusic.wav files from flash depending on situation.

Call Matches following outbound sip voip dial-peer on the ingress-gw

dial-peer voice 1400 voip
translation-profile incoming block
preference 1
max-conn 2
destination-pattern 5417641400
session protocol sipv2
session target ipv4:10.4.33.97 ## CUPS SIP PROXY SERVER IP ADDRESS
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad

CUPS load balance the call because there are static routes configured in it and sends call to CVP Call Server

Destination         VP Server IP Address         Priority           Weight  Protocol      
 5417641400          10.4.33.131 5060                1              1       UDP
 5417641400          10.4.33.132 5060                1              1       UDP 
 5417641400          10.4.33.133 5060                1              1       UDP

(2) CUPS —-> CVP Server

Call lands to the CVP Call SERVER and in the CVP Call Server log you could see following

[CVP Server Logs]

3052: 10.4.33.131: Mar 13 2008 14:34:39.703 -0700: %CVP_7_0_ICM-7-CALL: {Thrd=Pool:ICM[1450]} 
CALLGUID =  8B82AD85F07B11DC80250013192D1650, DLGID = 21 [SIP_LEG] - Publishing ,,    
[ICM_NEW_CALL],dialogueId=21,   sendSeqNo=1,trunkGroupId=100,trunkNumber=0,serviceId=1,
dialedNumber=5417641400,uui=,, LEGID = 8C1C7CEE-F07B11DC-8177F66C-9B680D18,DNIS = 5417641400,ANI = 4085274003

(3) CVP Server —> CUICME via VRU-PG

CVP Server sends the call to CUICM via VRU PG. Notice that we didn’t configure the IP address of the CUICM or the VRU-PG in the CVP Call Server. CVP Call Server automatically sends the information to the CUICM’s VRU-PG, because during the CUICM configuration, we had already specified the IP addresses of our CVP Call Servers. The VRU PG will connect to the ICM Service on the CVP Call Server. Once that TCP socket is established, the CVP acting as the routing client functionality becomes active in ICM and CVP will communicate with ICM/PG over that socket.

(4) CUICM [Runs Routing Script]—>Generates Label+Correlation-ID

When call arrives at CUICM, it causes Unified ICM to run a routing script. In the routing script there is a node called SendToVRU node (Remember: VRU in our case is VXML-GW and SendToVRU node is added right after setting the basic variables). The SendToVRU is what triggers ICM to transfer the call to the VXML GW (which is the VRU). The routing script will typically initiate a transfer of the call to a VoiceXML Gateway port (due to SendToVRU node) via the SIP Service. You could see following in the router process windows on the CUICM Server

[CUICME Router Logs]
14:51:07 Trace: CT_SL_Timer(56)(Init): CT(5000,PreSales_CT), Threshold(20 + 2).
14:51:07 Trace: Correlation id for dialog (23 x 0 : 0 0) is (44).
14:51:07 Trace: Dialog (23 x 44 : 0 0) transfer label = 5417641401

As you can see from above that ICM returns a VRU label 5417641401+correlation-id (44) back to CVP Call Server (Remember CVP Server was the routing-client).You can see the same information on the CVP Call Server logs that a call goes to ICM and ICM returns a label back to the CVP Call Server (Which is the routing-client for ICM). This correlation-id (which is 44 in our case ) will be appended at the end of the label.

[CVP Call Server Logs]
3053: 10.4.33.131: Mar 13 2008 14:34:39.718 -0700: %CVP_7_0_ICM-7-CALL: {Thrd=Pool:ICM[1451]} 
CALLGUID = 8B82AD85F07B11DC80250013192D1650, DLGID = 21 [SIP_LEG] - Processing ,,  
[ICM_TEMPORARY_CONNECT],   dialogueId=21,sendSeqNo=1,   label=5417641401,   correlationId=44,, 
LEGID = 8C1C7CEE-F07B11DC-8177F66C-9B680D18, DNIS = 5417641400, ANI = 4085274003

Now CVP Call Server will send this VRU label (5417641401+44) to the VXML GW (It is more accurate to say that CVP transfers the call TO the VXML GW, with the label+correlation ID as the destination phone number). In our case, it is only coincidence that the VXML GW is the same as the ingress GW. The “transfer” that CVP does is not a typical one either, because CVP stays in the call path (whereas usually the transferor drops out of the call path).

In reality CVP is setting up a NEW call to the VXML GW and is shuffling the SDP from the ingress GW to the VXML GW in order to connect the media directly (similar to CUCM or CUBE in flow-around mode). CVP stays in the signaling path so that it can retain call control.

[VXML-GW Debugs]

debug voip application vxml
calling Number=sip:4085274003@10.4.33.131:5060,(Calling Name=)(TON=National, NPI=ISDN, 
Screening=User, Passed, Presentation=Allowed),
Called Number=sip:541764140144@10.4.33.1:5060;transport=tcp(TON=Unknown, NPI=Unknown)

This label matches 1401 voip dial-peer in the VXML-GW and invokes bootstrap VXML application.

dial-peer voice 1401 voip

translation-profile incoming block
service bootstrap
session protocol sipv2
incoming called-number 5417641401T           ## T is used to match the correlation-id
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad

The bootstrap VXML application tells the CVP that it is a new call and CVP then sends the same label (5417641401+44) back to ICM. Once ICM sees its own generated label, this actually completes the loop and ICM at this point knows that there is a VXML Gateway ready to accept the instructions.

(5) CUICM—> RunExternalScript node

The steps below are the RunExternalScript nodes that tell CVP what VXML pages to send to the VXML gateway (play prompt, get digits, etc). CUICM execute the script instruction called “RunExternalScript”. This basically tells CVP that IVR related instructions are coming. CUICM then sends instructions like for example “Get4Digits”. CVP sends this instruction to VXML-GW to play the prompt to the caller. When caller enters the digit, ICM stores it in its variable [for example Call.CallerEnteredDigits variable].

[VXML-GW Debugs]
063686: Mar 13 18:25:14.383 PST: //404/vapp_digit_collection_done: digits [5432], status [0], pattern [v0]
063687: Mar 13 18:25:14.383 PST: //404/vxml_digit_collection_done:
  vxmlp 6A6D1EA4 status 0 async_status 200000000
063688: Mar 13 18:25:14.383 PST: //404/vxml_digit_collection_done: digits (5432)

ICM Script then instructs CVP to play the prompts back to the user on the PSTN side. For simplicity we can say that in order to do that CVP makes an inbound call to the VXML gateway by dialing 987654 and plays the prompt. But in reality we wouldn’t say that CVP makes an inbound call to the VXML gateway (since that already happened above). The 987654 is actually a transfer instruction that was received via VXML from CVP, the only reason it exists is because it is a workaround for some problem with playing media files and retaining the vxml session.

[VXML-GW DEBUGS]
063767: Mar 13 18:25:14.407 PST: //404/vxml_item_attrs_proc:
  name=mycall
  URI(abs):phone://987654
  scheme=phone
  host=987654 bridge=1 connecttimeout=0 maxtime=0 desttype=-1 destplan=-1 
  anitype=-1 aniplan=-1 anipi=-1, anisi=-1 rdntype=-1 rdnplan=-1 rdnpi=-1, rdnsi=-1, redirectreason=-1

063779: Mar 13 18:25:14.407 PST: //404/vapp_media_play: prompt=http://10.4.33.130/en-us/sys/5.wav:
063780: Mar 13 18:25:14.407 PST: //404/vapp_media_play: prompt=http://10.4.33.130/en-us/sys/4.wav:
063781: Mar 13 18:25:14.407 PST: //404/vapp_media_play: prompt=http://10.4.33.130/en-us/sys/3.wav:
063782: Mar 13 18:25:14.407 PST: //404/vapp_media_play: prompt=http://10.4.33.130/en-us/sys/2.wav:

(6) CUICM—->Agent

  • When an agent becomes available, ICM instructs CVP to transfer the call to the agent.
  • CVP will disconnect the call to the VXML GW (SIP BYE), will send a re-INVITE back to the ingress gateway to stop media, and will send a new INVITE to the agent phone and to the ringtone service below.
  • When the ringtone service answers, a re-INVITE is sent to the ingress gateway with the VXML GW’s SDP in order to establish the media stream for the ringback.
  • Once the Agent answers the phone, CVP disconnects the call to the VXML GW (with BYE & re-INVITE same as before), and then connects the media stream from the ingress gateway to the agent phone.
  • In this entire call flow, there have been 4 distinct SIP phone calls that are separate from each other:
    • ingress gateway to CVP;
    • CVP to VXML GW (IVR);
    • CVP to VXML GW (ringtone);
    • CVP to Agent.
  • Each phone call, from a SIP perspective, is completely separate from the others, because CVP is a B2BUA and not a proxy.
  • Whenever CVP does a transfer, it disconnects the previous destination that the caller was connected to and connects them to the new destination.
  • Thus, CVP acts as a pivot point for the call.
  • Since CVP is under control of ICM, that is what makes CVP so powerful – the ability to control the call from very first component that received the call. This is in contrast to a typical PBX/ACD (including CUCM) where the transfers are tromboned through every component that was involved in the call, tying up additional DS0’s or memory.
  • In a SIP call flow, during the time when CUICM is transferring call over to the agent, there is a need to play ringtone back to the caller. And also if in case there is an error during transfer, CVP should be able to play error tone back to the caller. These tones will be generated by playing a .wav file off of the VXML gateway flash.
  • In the CVP, there are two parameters under SIP section. These are the DNs that CVP sends to VXML GW when there is a need to generate the ringtone or error tone.
  • When these numbers reach the VXML GW, it matches with an incoming voip dial-peer and VXML GW starts the respective service/app
dial-peer voice 9191 voip
service ringtone
incoming called-number 91919191
dtmf-relay rtp-nte h245-signal h245-alphanumeric
codec g711ulaw
session protocol sipv2
no vad
!
dial-peer voice 9292 voip
service cvperror
incoming called-number 92929292
dtmf-relay rtp-nte h245-signal h245-alphanumeric
codec g711ulaw 
session protocol sipv2
no vad
  • You can see the VXML-GW playing the file by noticing the following debugs
[Debugs VXM GW]
062151: Mar 13 18:14:12.483 PST: //383/tcl_MediaObjCmd:  media play leg_incoming flash:ringback.wav
062152: Mar 13 18:14:12.483 PST: //383/tcl_MediaPlayObjCmd:  play leg_incoming flash:ringback.wav

NOTE: Ignore the timestamp in the debug messages. These debugs were captured by making multiple calls over a period of time

Different Gateway Applications

bootstrap.tcl

The script extracts required parameters and then hands off the call to the boot strap VXML service named “new-call” which loads the bootstrap.vxml file. This VXML page then submits a new call HTTP request to the Customer Voice Portal (CVP) IVR Service which kicks off a sequence of VXML communications between the VXML Gateway and the CVP IVR Service, commonly called “MicroApps”. For SIP calls, this script extracts Call GUID, Call Server URL, and DNIS. Call Server host is extracted from the App-Info header. For H.323 calls, Call GUID and DNIS are extracted. Call Server is also extracted from the call data. For both SIP and H.323 calls, the Call GUID is normalized in base16, 35 char length hex format as follows(not base10, 43 char length):
HHHHHHHH-HHHHHHHH-HHHHHHHH-HHHHHHHH

bootstrap.vxml

The VXML-GW sends a new call request to the Customer Voice Portal (CVP) IVR Service which kicks off a sequence of VXML communications between the VXML Gateway and the CVP IVR Service, commonly called “MicroApps”. Any HTTP errors, or fetch errors (7 second timeout), will handoff to recovery.vxml in flash and handoff.tcl in flash.

Handoff.tcl

Handoff.tcl has the sole job of disconnecting a call with a code an ISDN Q.850 code of 38.

survivablity.tcl

This service should be placed on the ingress-gw on the incoming pots dial-peer that is destined for CVP. In the event of critical CVP application errors or a WAN failure that would normally disconnect the caller, this script allows the gateway to attempt a transfer to some alternate location after the failure occurs instead of disconnecting the caller. In the event that the call cannot be transfered to an alternate agent, the script will play a “call-back-later” message and disconnect.

By VKN

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