Cisco Mobile Collaboration Solutionsd2zmdbbm9feqrf.cloudfront.net/2015/usa/pdf/CCSCOL-2650.pdf · TCL Script on Cisco UBE SIP INVITE for Inbound call to SMI handset SIP INVITE for

Cisco Mobile Collaboration Solutions

Sriram Sivaramakrishnan, Network Consulting Engineer

CCSCOL-2650

• Introduction to Mobile Collaboration Solutions

• Service Provider Mobile Integration Overview

• Service Provider Mobile Integration Call Flows

• Customer Requirements

• UC Solution

• Troubleshooting issues faced in SMI setup

• Q and A

Agenda

• What’s Fixed Mobile Convergence (FMC) and Fixed Mobile Substitution (FMS) ?

• How does “Forced-On” PBX calls work ?

• How to leverage UC components (Cisco Unified Communications Manager, Cisco Unity Connection, Cisco Unified Border Element, Cisco MediaSense) and Scripting (TCL/Python) to accomplish customer requirements ?

• Advantages of using FMC solution

Session Objectives

Customer Overview

• Customer has remote users who don’t have IP Phones registered to Cisco Unified CM environment

• Remote users use mobile smartphone for all “office” calls

• Calls currently routed and handled by the Telco

Overview Of Current Landscape

Lack of Management, Recording and Reporting for Mobile Users

• Technical Issue: No Jabber on Mobile Device

• User/Management Preference

• UC Connectivity Issues/Security Policies

• Impact to Customer

• Higher cost call activity

• Unable to record calls → Loss of Quality Assurance, Compliance

• Separate call reports → Additional Costs for Compiling and Correlating

• Device-stored Voicemails → Lack of Reporting and Backup

Customer’s Business Issues

Mobile Collaboration SolutionsFixed Mobile Convergence (FMC): Linking User's Enterprise and Mobile

What is fixed mobile convergence?

Integrating or linking a user's mobile device with their enterprise line

Enterprise Line

Mobile Voice

Network

(Carrier)

PSTN

Mobile Device

Mobile Collaboration Solutions

Fixed Mobile Substitution (FMS): Enabling User's Enterprise Line on Mobile Device

What is fixed mobile substitution?

Enterprise Line

Moving or enabling the enterprise line on the user's mobile device

Mobile Voice

Network

(Carrier)

PSTN

Mobile Device

Service Provider Mobile Integration Overview• Service Provider Mobile Integration (SMI) integrates the cell phone (handset) to the

Cisco UC infrastructure in the back end

• Service Provider intercepts call from/to SMI handset and sends it to Cisco UBE

• Cisco UBE integrates with Cisco Unified CM to handle it as a VoIP call

• Sprint solution :

http://shop.sprint.com/mysprint/services_solutions/details.jsp?detId=mobile_integration&catId=solution_ip_convergence_mobility&catName=IP%2520WAN%2520Convergence%2520-%2520Mobility&detName=Mobile%2520Integration

Considerations:

Software/application required on the mobile

device. What device(s) do I need?

WLAN or Mobile Data or both?

Substitution: Do I need an enterprise device?

Answers:

None. Handsets enabled for Provider Mobile

Integration

Connectivity to telco network

No enterprise device is needed on Cisco Unified CM

http://shop.sprint.com/mysprint/services_solutions/details.jsp?detId=mobile_integration&catId=solution_ip_convergence_mobility&catName=IP WAN Convergence - Mobility&detName=Mobile Integration

SMI Call Flows

SMI Call Flows – Inbound Call To SMI

TelcoIP Cloud

PSTN CallerUnified Mobility SNR

SMI Handset

RTP Stream

SMI Call Flows – Outbound Call From SMI

TelcoIP Cloud

PSTN phone

SMI Handset

RTP Stream

SMI Call Flows – Call From SMI To VM Box

TelcoIP Cloud

SMI Handset

RTP Stream

Unity

Cxn

Solution Overview

Customer Requirements

• Recording Prompt for Incoming Calls

• Outbound SMI Calls shouldn’t be recorded

• 25% recording of Incoming Calls

• Cisco Unity Cxn Voicemail boxes for SMI Handsets

• Voicemail Notification for SMI Handsets

• Cisco Unified CM/Cisco Unity Cxn scheduled reporting for SMI Handsets








TCL Script on Cisco UBESIP INVITE for Inbound call to SMI handset SIP INVITE for Outbound call from SMI handset

Received:

INVITE sip:[email protected];user=phone;transport=udp SIP/2.0

To: <sip:[email protected];user=phone>

From: <sip:[email protected]:51040;isup-

oli=61;user=phone>;tag=SDopffd01-127.0.0.1alUtKGp-01044+1+851f006e+16e92d5e

Call-ID: SDopffd01-329fe6994e4cc363069bccef5736419c-c54i532

CSeq: 270039797 INVITE

Max-Forwards: 40

Content-Length: 231

Contact: <sip:[email protected]:5060;transport=udp>

Content-Type: application/sdp

Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY,

PRACK, INFO, REFER, UPDATE, PUBLISH, MESSAGE

Supported: replaces, timer

P-Asserted-Identity: <sip:[email protected];user=phone>

Expires: 180

Min-SE: 1800

Session-Expires: 1800

Request-Disposition: fork, parallel

Privacy: none

User-Agent: X

P-Charging-Vector: icid-value=032d4c720b58d645c0b398ebd14abbf4

P-Charging-Function-Addresses: ccf="aaa://telco.spectrum.com:3868;transport=tcp"

P-Served-User: sip:[email protected];sescase=term;regstate=unreg

Route: sip:[email protected]:5060;user=phone;lr

<SDP>

Received:

INVITE sip:[email protected];user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK43o02s105891lj8hn681.1

To: <sip:[email protected];user=phone>

From: <sip:[email protected]:51040;isup-oli=61;user=phone>;tag=SDk6eb501-

127.0.0.1alUtKGp-01042+1+b09c007d+7ce5d92d

Call-ID: SDk6eb501-5529cf273e94e0e3dead001098fc16ba-c54i532

CSeq: 871042177 INVITE

Max-Forwards: 58

Content-Length: 231

Contact: <sip:[email protected]:5060;transport=udp>

Content-Type: application/sdp

Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK,

INFO, REFER, UPDATE, PUBLISH, MESSAGE

Supported: replaces, timer

P-Asserted-Identity: <sip:[email protected];user=phone>

Expires: 180

Min-SE: 1800

Session-Expires: 1800

Request-Disposition: fork, parallel

Privacy: none

User-Agent: X

P-Charging-Vector: icid-value=8a255ede0b5f28c100b3985af13c0781;orig-

ioi=telco.domain.net

P-Charging-Function-Addresses: ccf="aaa://telco.spectrum.com:3868;transport=tcp"

P-Served-User: sip:[email protected];sescase=orig;regstate=unreg

Route: <sip:[email protected]:5060;user=phone;lr>

<SDP>

sip:[email protected]:5060;user=phone;lr








Media-Forking

Telco IP Cloud

SNR

SMI Handset Call’s RTP Stream

Cisco UBE media

forked stream

PSTN Caller

Record Percentage Of Calls ?

• Multiple Cisco UBE outbound dial-peers with same preference

• Some enabled for media-forking and the rest not set up for media-forking !

• Ensure circular dial-peer hunting on Cisco UBE (‘dial-peer hunt 1’)








Cisco Unified CM / Cisco Unity Cxn design

• Single Number Reach (SNR) on Cisco Unified CM

• Remote Destination (RD)

• Delay Before Ringing Timer = 0

• Answer Too Soon Timer = 0

• Answer Too Late Timer = 23500 msec (23.5 seconds)

• Maximum Wait Time for Desk Pickup on SNR User = 0 seconds

• Voicemail disabled by Telco on the local SMI handsets

• Voicemail boxes created on Cisco Unity Cxn for the SMI numbers








MWI For SMI user ?

SMTP Smart Host (Relay)

SMI Handset Unity Cxn<SMI_Number>@telco.messaging.com

TelcoSMS

• MWI Disabled for Cisco Unity Cxn User

SMI Supported Features







Reporting Requirements

• Avg # of Calls Handled/Placed per SMI Handset ?

• # of Incoming Calls to SMI Handsets rolling over to voicemail ?

• Avg # of Voicemails left for SMI Handsets ?

• Promptness of SMI Users in checking their voicemails ?

• Promptness of SMI Users in returning calls based on voicemails ?







Cisco Unified CM Reporting

• Call Detail Records (CDR) !

• Configure external server to push raw CDR files out to, from the CDR Repository (Cisco Unified CM Publisher)







Python Script Host (PSH)

Schedule Script On PSH

Email Option on PSH














Troubleshooting

Some Troubleshooting Pointers

Scenario Underlying Issue Resolution

Custom call handling None TCL script on Cisco UBE

SMI user tries to answer call

immediately when call rings, but

call is disconnected

None. Single Number Reach

functionality to prevent call from

going to cell phone’s voicemail

Voicemail on SMI cell phones

are disabled by Telco – so, set

the ‘Answer Too Soon’ timer on

Remote Destination (RD) config

from default 1500 msec to 0

msec

If SMI user hangs up, caller

hears Music on Hold for a few

seconds and call disconnects

None. Single Number Reach

functionality to transfer call from

cell phone (RD) to desk phone

Cisco Unified CM Administration

> User Management > End

User > set the ‘Maximum Wait

Time for Desk Pickup’ field

from default 10000 to 0

Bulk disabling MWI notification

for SMI users

Bulk Edit from Cisco Unity Cxn

GUI is not an option

Python Script !

Solution Review

FMC bridges the gap between a company’s fixed and mobile workforces

Leverages existing Cisco UC Investment

Forces mobile calls ‘On-Net’ for Monitoring, Recording and Reporting

Custom Call Handling for Mobile Users

Centralized Voicemail

FMC Review

• What problems are we solving for the customer ?

• Alternative to Jabber-on-mobile

• Platform to leverage UC infrastructure and provide custom reports

• Advantages to the customer• Cisco Mobility to integrate Mobile and Desk phone

• Reduced costs

• Better able to understand the customer experience

• Enabling better customer service

The Customer Opportunity

• Challenges with the solution ?

• 911 call from SMI phones will not be routed through UC setup

• More points of failure (Telco / UC)

• MWI is an SMS

• Careful planning of Cisco UBE / MediaSense capacity

Caveats

• Are the services specific to any geography?

• FMC generally is a global feature set

• Unified Mobility on Cisco Unified CM universally available

• Case study is US-based

• Sprint and AT&T – service providers used by customer

• Providers outside the US also offer FMC/Forced-On-PBX services

Services

The Key Takeaways of this presentation were:

FMC/SMI to route all remote users’ calls through UC setup

Single Number Reach (SNR) on Cisco Unified CM to route calls to remote users

TCL script on Cisco UBE for custom call handling

Cisco UBE Media Forking and MediaSense to record calls

Cisco Unity Cxn for voicemails, MWI as SMS

Python Scripting Host (PSH) for custom Voicemail reports

Key Takeaways

Cisco Live San Diego Breakout Sessions

BRKUCC-2060 - Cisco Enterprise Mobile Collaboration (Tuesday, June 9, 8 AM – 10 AM)

Presenter – Matthew Jordy, TME

BRKUCC-2932 - Troubleshooting SIP with Cisco Unified Communications (Tuesday, June 9, 8 AM – 10 AM)

Presenter – Paul Giralt, DSE

BRKUCC-2934 - Implementation and Management of Cisco Unified Border Element(CUBE) Enterprise (Tuesday, June 9, 1 PM – 3 PM)

Presenters – Hussain Ali, TME and John Vickroy, PM

BRKUCC-2270 - Network Media Recording and Streaming with Cisco MediaSense(Wednesday, June 10, 8 AM – 10 AM)

Presenter – Christopher Ward, TME

BRKCCT-2050 – Building, recording and monitoring applications with the MediaSenseAPI (Wednesday, June 10, 1 PM – 3 PM)

Presenter – Ken Rehor, PM

Additional Resources

Summary

Business need for FMC :

Lack of Management, Recording and Reporting for Mobile Users

Technical Issue: No Jabber on Mobile Device

User/Management Preference

UC Connectivity Issues/Security Policies

Impact to Customer

Higher cost call activity

Unable to record calls → Loss of Quality Assurance, Compliance

Separate call reports → Additional Costs for Compiling and Correlating

Device-stored Voicemails → Lack of Reporting and Backup

Summary (1 of 3)

What problems are we solving for the customer ?

FMC bridges the gap between a company’s fixed and mobile workforces

Alternative to Jabber-on-mobile

Platform to leverage UC infrastructure and provide custom reports

Advantages to the customer Reduced costs

Better able to understand the customer experience

Enabling better customer service

Summary (2 of 3)

• SMI Supported Features

• Recording Prompt for Incoming Calls → TCL script on Cisco UBE

• Outbound SMI Calls shouldn’t be recorded → TCL script on Cisco UBE

• 25% recording of Incoming Calls → Dial-peers on Cisco UBE,Media Forking, MediaSense

• Unity Cxn Voicemail boxes for SMI Handsets → Unity Cxn VM / SNR on

Cisco Unified CM

• Voicemail Notification for SMI Handsets → SMTP notification

• Cisco UCM scheduled reporting for SMI Handsets → Cisco Unified CM CDRs

• Unity Cxn scheduled reporting for SMI Handsets → Python Scripting Host (PSH)

Summary (3 of 3)

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Thank you

Supporting Slides

SMI Call Flow - Overview

• Telco intercepts call from/to SMI handset and sends it to CUBE

• CUBE routes call to CUCM

• For calls from SMI Handset, CUCM then routes the call based on the dial plan

• For calls intended to SMI Handset, CUCM uses Single Number Reach (SNR) – RDP/RD conceptto route the call back to CUBE and Telco IP cloud.

• SIP OPTIONS PING between CUBE and Telco, to ensure SIP connectivity between Telco andCisco UC infrastructure.

• Signaling is all SIP !

• Signaling “anchored” through CUCM

• Media flow-through on CUBE (RTP media “anchored” through CUBE)

SMI Call Flows - Review• Call to SMI Handset

• PSTN phone Telco IP Cloud CUBE CUCM RDP/RD CUBE Telco IP Cloud SMI Handset

• Call from SMI Handset

• SMI Handset Telco IP Cloud CUBE CUCM Dial Plan Egress GW / internal number

• Call from SMI Handset to Voicemail

• SMI Handset Telco IP Cloud CUBE CUCM TP Voicemail pilot SIP/SCCP to CUC

SMI – Digit Prefix

• How would CUBE know it’s an SMI-based call or a regular call ?

• Telco adds a prefix (467) to every SMI-based call (from/to SMI handset) and sends it to CUBE

• How would Telco know to send the call to SMI Handset when call is hairpinned through CUBE ?

• CUCM changes the prefix (99999) for call to SMI handset and sends it to Telco IP cloud. Once Telco sees the prefix it is expecting (99999), it will route the call to the SMI handset

Inbound call to SMI - Example

TelcoIP Cloud

PSTN Caller

SNRRDP – 4679195740870RD – 999999195740870

SMI Handset9195740870

RTP Stream

SMI number being called - 9195740870

Called number :

9195740870

Called number :

4679195740870

Called number :

999999195740870

Called number :

999999195740870

Called number :

4679195740870

Strip 99999 prefix and

send to SMI handset

Outbound call from SMI - Example

TelcoIP Cloud

PSTN phone 4085551234

SMI Handset

RTP Stream

Called number :

4085551234

Called number :

4674085551234

Called number :

4674085551234

Called number :

4085551234

Strip 467, append 91 and

route based on existing Dial

Plan (TP, RP)

SMI Handset to VM box - Example

TelcoIP Cloud

SMI Handset

RTP Stream

CUC

Called number :

9193921111

VM Pilot number – 9193921111

Called number :

4679193921111

Called number :

4679193921111

Strip 467, send to VM pilot

number

TCL Scripting on CUBE

Why TCL ?

Provides call-handling options

Configured on the inbound dial-peer on CUBE, matched when call comes in from Telco IP Cloud

Config :

application

service smi flash:smirecording.tcl

dial-peer voice 101 voip

description “Inbound dial-peer with recording prompt”

service smi

incoming called-number <smi-number>

TCL Enhancements

Depending on the direction of the call wrt SMI handset, TCL script can handle and route the call differently

Outbound Call from SMI Handset : P-Charging-Vector field has “orig-ioi=telco.domain.net”

Inbound Call for non-SMI Handset : P-Charging-Vector field does not have “orig-ioi=telco.domain.net”

TCL Script

Accept incoming call

Extract P-Charging-Vector and Called Party Number from SIP INVITE message

If P-Charging-Vector field has “orig-ioi=telco.domain.net” (outbound call from SMI Handset)

Do not play recorded prompt

Send call to CUCM

If P-Charging-Vector field does not have “orig-ioi=telco.domain.net” (call from non-SMI Handset)

Play recorded prompt

Send call to CUCM

Header-passingPrior to 15.x IOS version, voice applications running on CUBE did not have access to headers sent in SIP

requests

SIP Header Support introduces a new command, the header-passing command, to either enable or

disable passing headers from INVITE messages to applications.

The SIP Header Passing feature also provides enhanced inbound and outbound dial-peer matching

services

Config :

voice service voip

sip

header-passing

header-passing enabled header-passing disabled

‘set calledparty [infotag get

leg_dnis]’ returns the SIP URI

‘set calledparty [infotag get leg_dnis]’ returns the

called party number

Enhanced Scriptproc init { } {

global param

global calledparty

set param(interruptPrompt) true

set param(abortKey) *

set param(terminationKey) #

set param(maxDigits) 12

set param(interDigitTimeout) 10

}

proc act_Accept_Inbound_Call_Leg { } {

global calledparty

set legdn [infotag get leg_dn_tag]

puts "leg dn is $legdn"

set dnisuri [infotag get leg_dnis]

puts "original called party uri is $dnisuri"

regexp {sip:(.*)@.*} $dnisuri matchall calledparty

puts "matchall is $matchall"

puts "called party is $calledparty"

set vector [infotag get leg_proto_headers P-Charging-Vector]

puts "vector is $vector"

leg setupack leg_incoming

leg proceeding leg_incoming

leg connect leg_incoming

if {[regexp {orig-ioi=telco.domain.net} $vector] == 1} then {

timer start leg_time 1 leg_incoming

} else {

media play leg_incoming "flash:/RecordingWarning.wav“

regsub {467} $calledparty 555 calledparty

puts "New called number is $calledparty"

}

}

proc act_Connect_Outbound_Call_Leg { } {

global calledparty

puts "Final called party is $calledparty"

leg setup $calledparty callinfo leg_incoming

}

proc act_Is_Call_Setup_Done { } {

set status [infotag get evt_status]

if { $status == "ls_000" } {

call active

} else {

puts "Call Connection Failure"

call close

}

}

proc act_Cleanup { } {

call close

}

init

#----------------------------------

# State Machine

#----------------------------------

set SMIsFSM(st_Call_Init,ev_disconnected) "act_Cleanup,st_Call_Disconnected"

set SMIsFSM(st_Call_Init,ev_setup_indication) "act_Accept_Inbound_Call_Leg,st_Call_Outbound_Call_Leg"

set SMIsFSM(st_Call_Outbound_Call_Leg,ev_media_done) "act_Connect_Outbound_Call_Leg,st_Call_Active"

set SMIsFSM(st_Call_Outbound_Call_Leg,ev_leg_timer) "act_Connect_Outbound_Call_Leg,st_Call_Active"

set SMIsFSM(st_Call_Active,ev_disconnected) "act_Is_Call_Setup_Done,st_Call_Disconnected"

set SMIsFSM(st_Call_Disconnected,ev_disconnected) "act_Cleanup,same_state"

set SMIsFSM(st_Call_Disconnected,ev_media_done) "act_Cleanup,same_state"

set SMIsFSM(st_Call_Disconnected,ev_disconnect_done) "act_Cleanup,same_state"

set SMIsFSM(st_Call_Disconnected,ev_leg_timer) "act_Cleanup,same_state"

fsm define SMIsFSM st_Call_Init

(contd)..

Clipped Recording Prompt

But ….

The recording prompt played from CUBE was clipped for the calling party. For example, if the recorded prompt was “Your call might be recorded”, the calling party heard “might be recorded” when calling a SMI handset

From packet captures from CUBE, CUBE was playing the prompt fully but the RTP channel wasn’t open all the way through to the calling party

Even though it doesn’t seem to be a CUBE issue, can we remediate the issue from CUBE side ?

YES !

Introduce a delay between the Connect and the Recording Prompt being played

Failure CallsHow to handle calls if

SMI dialed an invalid / inaccessible number ?

Mobile user dialed an SMI extension which is not yet configured on the CUCM ?

SMI dialed an invalid / inaccessible number ?

Do not connect the incoming call» if {[regexp {orig-ioi=telco.domain.net} $vector] == 1} then {

» timer start leg_time 1 leg_incoming

» }

If CUBE gets 404 Not Found from CUCM and hunting options have been exhausted, event ‘ev_setup_done’ is generated.

Separate functions for calls from SMI and calls to SMI» proc act_Is_Call_Setup_Done { }

» proc act_Is_Call_to_SMI_Setup_Done { }

Separate FSM states for calls from SMI and calls to SMI» set TopFSM(st_Play_Prompt,ev_media_done) "act_Connect_Outbound_Call_Leg,st_Call_to_SMI"

» set TopFSM(st_Call_Active,ev_setup_done) "act_Is_Call_Setup_Done,st_Call_Disconnected"

» set TopFSM(st_Call_to_SMI,ev_setup_done) "act_Is_Call_to_SMI_Setup_Done,st_Call_Disconnected"

Failure Calls (contd..)

Mobile user dialed an SMI extension which fails ?

Play message that the call failed•proc act_Is_Call_to_SMI_Setup_Done { } {

• set status [infotag get evt_status]

• puts "SMI Call Status is $status"

• if { $status == "ls_000" } {

• puts "In SMI IF Condition"

• puts "The call was successfully setup"

• } else {

• puts "Call to SMI Connection Failure"

• media play leg_incoming "flash:/Error_Prompt.wav"

• }

•}

FSM state for this scenario

• set TopFSM(st_Call_to_SMI,ev_setup_done) "act_Is_Call_to_SMI_Setup_Done,st_Call_Disconnected“

• set TopFSM(st_Call_Disconnected,ev_disconnected) "act_Cleanup,same_state"

• set TopFSM(st_Call_Disconnected,ev_disconnect_done) "act_Cleanup,same_state"

Final TCL Scriptproc init { } {

global param

global calledparty

global matchall

set param(interruptPrompt) true

set param(abortKey) *

set param(terminationKey) #

set param(maxDigits) 12

set param(interDigitTimeout) 10

}

proc act_Accept_Inbound_Call_Leg { } {

global calledparty

global matchall

set legdn [infotag get leg_dn_tag]

puts "leg dn is $legdn"

set dnisuri [infotag get leg_dnis]

puts "original called party uri is $dnisuri"

regexp {sip:(.*)@.*} $dnisuri matchall calledparty

puts "matchall is $matchall"

puts "called party is $calledparty"

set vector [infotag get leg_proto_headers P-Charging-Vector]

puts "vector is $vector"

leg setupack leg_incoming

leg proceeding leg_incoming

if {[regexp {orig-ioi=telco.domain.net} $vector] == 1} then {

timer start leg_time 1 leg_incoming

} else {

leg alert leg_incoming

leg connect leg_incoming

regsub {467} $calledparty 555 calledparty

puts "New called number is $calledparty"

puts "Starting 1 second DELAY_TIMER"

timer start named_timer 1 DELAY_TIMER

}

}

proc act_Play_Prompt { } {

puts "Going to play media that call will be recorded"

media play leg_incoming "flash:/RecordingWarning.wav"

}

proc act_Connect_Outbound_Call_Leg { } {

global calledparty

puts "Final called party is $calledparty"

leg setup $calledparty callinfo leg_incoming

}

proc act_Is_Call_Setup_Done { } {


puts "Status is $status"

if { $status == "ls_000" } {

puts "The call was successfully setup"

} else {

puts "Call Connection Failure"

call close }

}

proc act_Is_Call_to_SMI_Setup_Done { } {


puts "SMI Call Status is $status"

if { $status == "ls_000" } {

puts "The call was successfully setup"

} else {

puts "Call to SMI Connection Failure“

media play leg_incoming "flash:/Error_Prompt.wav“ }

}

proc act_Cleanup { } {

call close

}

Init

#--State Machine--------------------------------

set TopFSM(st_Call_Init,ev_disconnected) "act_Cleanup,st_Call_Disconnected"

set TopFSM(st_Call_Init,ev_setup_indication) "act_Accept_Inbound_Call_Leg,st_Call_Outbound_Call_Leg"

set TopFSM(st_Call_Outbound_Call_Leg,ev_leg_timer) "act_Connect_Outbound_Call_Leg,st_Call_Active"

set TopFSM(st_Call_Outbound_Call_Leg,ev_named_timer) "act_Play_Prompt,st_Play_Prompt"

set TopFSM(st_Play_Prompt,ev_media_done) "act_Connect_Outbound_Call_Leg,st_Call_to_SMI"

set TopFSM(st_Call_Active,ev_disconnected) "act_Is_Call_Setup_Done,st_Call_Disconnected"

set TopFSM(st_Call_Active,ev_setup_done) "act_Is_Call_Setup_Done,st_Call_Disconnected"

set TopFSM(st_Call_to_SMI,ev_setup_done) "act_Is_Call_to_SMI_Setup_Done,st_Call_Disconnected"

set TopFSM(st_Call_Disconnected,ev_media_done) "act_Cleanup,same_state"

set TopFSM(st_Call_Disconnected,ev_disconnected) "act_Cleanup,same_state"

set TopFSM(st_Call_Disconnected,ev_disconnect_done) "act_Cleanup,same_state"

set TopFSM(st_Call_Disconnected,ev_leg_timer) "act_Cleanup,same_state"

fsm define TopFSM st_Call_Init

(contd)..

MediaSense and CUBE Media Forking

Cisco MediaSense• Cisco’s recording solution

• SIP signaling to CUBE / CUCM

• Record both RTP streams (customer RTP stream and agent RTP stream) of the call

• Options :

• MediaSense SRE node

• MediaSense Primary/Secondary node

• MediaSense Expansion node

• Max 5 MediaSense nodes in a cluster – Primary, Secondary and 3 Expansion nodes

• Each MediaSense node (Primary/Secondary/Expansion) can store up to 12 TB of recorded media

• Recorded media is not replicated across MediaSense nodes in the cluster

• MediaSense node handling SIP signaling will direct RTP stream to a different MediaSense node if its storage space is getting full

CUBE Media Forking

• CUBE can fork media to recording server

• Can be used in a network-based recording solution to fork active SIP-SIP calls traversing through it.

• Feature supported from Cisco IOS release 15.2(1)T and later

• Supports Cisco’s Open Recording Architecture (ORA)

• CUBE at the network layer would act as a Recording Client

• Cisco MediaSense will act as a Recording Server at the Services Layer

CUBE Config


description "Dialpeer pointing to the MediaSense primary recording server"

destination-pattern 99999

session protocol sipv2

session target ipv4:X.X.X.X

session transport tcp

!


description "Dialpeer pointing to the MediaSense secondary recording server"

preference 1

destination-pattern 99999



session transport tcp

media class 3

recorder profile 10000

!

media profile recorder 10000

media-recording 123456 123457

!


description "Call leg from CUBE to Telco to SMI User”

preference 1

destination-pattern ^99999[2-9]..[2-9]......$



voice-class sip profiles 764

voice-class sip bind control source-interface Loopback0

voice-class sip bind media source-interface Loopback0

media-class 3

dtmf-relay rtp-nte

codec g711ulaw

no vad

Record Percentage Of Calls ?


description "Call leg from CUBE to Telco to SMI User – media forking enabled"

preference 1



media-class 3

!


description "Call leg from CUBE to Telco to SMI User – no media forking"

preference 1




description "Call leg from CUBE to Telco to SMI User – no media forking"

preference 1



!


description "Call leg from CUBE to Telco to SMI User – no media forking”

preference 1



!

dial-peer hunt 1

*** dial-peer hunt enums in the Notes of Slide ***

Calls to SMI Supervisors shouldn’t be recorded ?

Solution : Configure specific outbound dial-peer for every SMI Supervisor. Do NOT enable media forking for these dial-peers !

Dial-peer 211 has no ‘media-class’ configuration on it. Calls to this SMI Supervisor will not be recorded.


description "Call leg from CUBE to Telco to SMI Supervisor"

huntstop

destination-pattern ^99999XXXXXXXXXX$



voice-class sip asserted-id pai

voice-class sip profiles 764

voice-class sip options-keepalive

voice-class sip bind control source-interface Loopback0

voice-class sip bind media source-interface Loopback0

dtmf-relay rtp-nte

codec g711ulaw

no vad

MWI For SMI user ?

• SMI user is remote to UC infrastructure, so traditional MWI notification will not work

• SMTP Notification on CUC for SMI user !

• ‘To’ address = <SMI_Number>@telco.messaging.com

• telco.messaging.com converts email into SMS for SMI phone number

• Message Text will be something generic like “New Voicemail has been received. Please dial 919-392-1111 to listen to it.”

• 919-392-1111 will be a hyperlink on SMI handset, to dial

• Disable MWI notification for the CUC users

Unity Connection Reporting

• From CUCM CDR, can see if call rolled over to voicemail

• However, no way to ascertain if voicemail was actually left or not

• Option :

• CUC Reports !

• But …

• No automated way from CUC GUI to schedule voicemail report generation

• No automated way from CUC GUI to export voicemail records

Solution :

• Python Scripting Host (PSH) for automating reports

Unity Connection Reporting - Python Scripting Host (PSH)

• http://www.ciscounitytools.com/Applications/CxN/PythonScriptingHost/PythonScriptingHost.html

• Uses CUC DB Proxy service to connect to the CUC database

• Write your own Python script to accomplish the task !

• For Voicemail reports :• Create CSV file with SMI user aliases

• Pass in CSV file to the script

• Connect to CUC Reporting Database (UnityRptDB)

• Pull the voicemail report for every SMI user in the CSV file

• Store the output in a file

• Run VM report on both CUC Pub and Sub, as Reporting DB is not replicated between Pub and Sub

http://www.ciscounitytools.com/Applications/CxN/PythonScriptingHost/PythonScriptingHost.html

How To Look At CUC DB Structure - CUDLI

Combining CUCM and CUC Reports

How prompt are the SMI agents in returning their voicemails ?

From PSH report, check calling and called party on Voicemail for SMI user

On CUCM CDR, swap the calling and called party numbers and look for call from SMI to caller who

left voicemail

Based on CDR times on PSH and CDR reports, calculate the time lag between time of voicemail

and time of call back => determines promptness of SMI agents in returning their voicemails

However …

A good but not foolproof measure of promptness

Voicemail might have mentioned a different number to call back on

Voicemail might have been a call which didn’t need a call back

In essence :

Combining CUCM and CUC reports provides leverage to customers

• Providers outside the US also offer FMC/Forced-On-PBX services.

• Vodafone One Net (http://www.vodafone.co.uk/business/products-and-services/unified-communication/one-net-business/one-net-business-4/index.htm)

• BT One (http://www.globalservices.bt.com/uk/en/products_category/unified_communications_services)

Services

http://www.vodafone.co.uk/business/products-and-services/unified-communication/one-net-business/one-net-business-4/index.htm

http://www.globalservices.bt.com/uk/en/products_category/unified_communications_services

More Troubleshooting PointersScenario Underlying Issue Resolution

How are 911 calls handled in SMI setup ? None 911 calls are routed directly through Telco

network. 911 calls are not sent to Cisco

UBE/Cisco Unified CM setup by telco.

Caveat – 911 calls from SMI users will not

be captured on Cisco Unified CM CDR.

Bulk modifying SMTP Message Text when a

new voicemail is received

Bulk Edit from Cisco Unity Cxn GUI is not an

option

Python Script !

Call to SMI number results in fast busy or

prompt “The number you are calling is

unavailable right now”

Call not reaching Cisco UC setup Troubleshoot connectivity between Telco

and CUBE

No VM Notification text messages received

by the SMI user

SMTP Notification from CUC via Telco

network did not reach the SMI phone

Troubleshoot SMTP connectivity from CUC

to SMTP Relay, and Telco to translate to

SMS

More Troubleshooting Pointers (contd…)Scenario Underlying Issue Resolution

Duplicate Call Detail Records (CDR) for

some calls to SMI that rolled over to VM

None, as per design CUBE receives 404 Not Found from Telco

and sends that to CUCM. CDR created for

failed call by design

Python Scripting Host not logging in to CUC

Database

CUC Database Proxy Service may have

stopped

Advanced > Connection Administration >

Database proxy: Service Shutdown Timer (in

Days) – 999

This value keeps decrementing every day.

Activate and Start Database Proxy Service

Database Proxy Service will be stopped if

the CUC server is rebooted. Remember

to restart Database Proxy Service after a

CUC server reboot.

Recording playback not working from Cisco

MediaSense Search and Play

Firewall Ports not opened Documentation bug “CSCuq13398 -

Recording Playback ports not documented

in MediaSense Port Usage table”

From tests,

- 16384-32767 (RTP) used as the RTP

source port from MediaSense servers to the

MediaPlayer

- 16384-65535 (RTP) will be used the RTP

destination port on the web browser running

the MediaPlayer

Documents

Cisco Mobile Collaboration Solutionsd2zmdbbm9feqrf.cloudfront.net/2015/usa/pdf/CCSCOL-2650.pdf · TCL Script on Cisco UBE SIP INVITE for Inbound call to SMI handset SIP INVITE for