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Document Overview

This document provides a configuration guide for Sonus SBC 1000/2000 series (Session Border Controller) when connecting to Cisco Unified Communications Manager 10.5 (CUCM 10.5) and CenturyLink SIP Trunk.

This configuration guide supports features given in the Cicso UCM configuration guide.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound call flows between Sonus SBC 1000/2000 series and Cisco Unified Communications Manager 10.5 (CUCM 10.5).

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC and the third-party product. There will be steps that require navigating third-party as well as the Sonus SBC Command Line Interface (CLI). Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are also necessary to complete the configuration and for troubleshooting, if necessary.

This configuration guide is offered as a convenience to Sonus customers. The specifications and information regarding the product in this guide are subject to change without notice. All statements, information, and recommendations in this guide are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information in this guide.

Requirements

The following equipment and software were used for the sample configuration provided:

 

Equipment

Software Version

Sonus NetworksSBC 2000 V5.0.3build407
Tenor AF P108-09-21
Third-Party EquipmentCisco Unified Communications Manager
10.5.1.11005-2
Cisco IP Phone 79429.3.1.57

Reference Configuration

The following reference configuration shows connectivity between third-party equipment and Sonus SBC 1000/2000 series.

Support

For any questions regarding this document, please contact your maintenance and support provider.

Technical support for Sonus SBC 1000/2000 series is available via phone or logging a trouble ticket.

 

Third-Party Product Features

The following third-party product features are supported:

  • Basic originated and terminated calls
  • Basic inbound/outbound call
  • Hold and Resume
  • Call Forwarding
  • FAX
  • DTMF
  • Conference Call
  • Action on Loss of primary SBC 

Verify License

No special licensing is required for this test.

CUCM 10.5 Configuration

The following new configurations are included in this section:

  1. Security Profile
  2. SIP Profile
  3. SIP Trunk
  4. Route Group
  5. Route List
  6. Route Pattern

1.Security Profile

Select System > Security > SIP Trunk Security Profile 

Figure : Security Profiles

 

 

 

2.SIP Profile

Select Device > Device Settings > SIP Profile

Figure : SIP Profile

 


3.SIP Trunk

Select Device > Trunk > Add New

Figure : Primary SIP Trunk

 

 

 

Figure : Secondary SIP Trunk

 

 

4.Route Group

Select Call Routing > Route/Hunt > Route Group > Add New

Figure : Route Group

  

5.Route List

Select Call Routing > Route/Hunt > Route List > Add New

Figure : Route List

 

6.Route Pattern

Select Call Routing > Route/Hunt > Route Pattern > Add New

Figure : Route Pattern

   

 

Sonus SBC 1000/2000 Series Configuration

The following configuration steps provide an example of how to configure the Sonus SBC 1000/2000 series to interoperate with CUCM10.5 and CenturyLink SIP Trunk:

  1. SIP Profiles
  2. SIP Server
  3. Media Profiles
  4. Media List
  5. Remote Authorization Tables
  6. Contact Registrant Tables
  7. Transformation
  8. Signaling Groups 
  9. Call Routing Table  
  10. Cause Code Reroute
  11. Condition Rule Table
  12. Message Rule Table

1. SIP Profiles

SIP Profiles control how the Sonus SBC 1000/2000 series communicates with SIP devices. These control important characteristics such as session timers, SIP Header customization, SIP timers, MIME payloads, and option tags. 

Select Settings > SIP > SIP Profiles to access the SIP Profile screen.

The default SIP profile used for the SBC 1000/2000 series for this testing effort is provided in the following figures:

Figure : CenturyLink SIP Profile

Figure : CUCM 10.5 SIP Profile


Figure : Fax SIP Profile


2. SIP Server

SIP Server Tables contain information about the SIP devices connected to the Sonus SBC 1000/2000 series. 

Select Settings > SIP > SIP Server Tables to access the SIP Server Tables screen.

The entries in the tables provide information about the IP addresses, ports, and protocols used to communicate with each SIP server. The entries also contain links to counters that are useful for troubleshooting, as shown in the following figures:

Figure : CenturyLink Primary SIP Server

Figure : CenturyLink Secondary SIP Server

Figure : CUCM 10.5 Primary SIP Server

Figure : CUCM 10.5 Secondary SIP Server


Figure : Fax SIP Server


3. Media Profiles

Media Profiles specify the individual voice and fax compression codecs and their associated settings for inclusion into a Media List. Different codecs provide varying levels of compression, allowing the reduction of bandwidth requirements at the expense of voice quality. 

Select Settings > Media > Media Profiles to access the Media Profiles screen.

The following figures show the media profiles of the voice codecs used for the SBC 1000/2000 series in this testing effort and are provide for reference only.

Figure : CenturyLink Codecs


Figure : CUCM 10.5 Codec


4. Media List

The Media List shows the selected voice and fax compression codecs and their associated settings. 

 Select Settings > Media > Media List to access the Media List configuration screen.

Figure : CenturyLink Media List

Figure : CUCM 10.5 Media List

Figure : Fax Media List

5. Remote Authorization Tables

Remote Authorization Tables and their entries contain information used to respond to request message challenges by an upstream server. The Remote Authorization Tables on this page appear as options in the Remote Authorization and Contacts Panel for SIP Servers.

Select Settings > SIP > Remote Authorization Tables to access the Remote Authorization Tables configuration screen.

Figure : CenturyLink Primary Trunk Table


Figure : CenturyLink Secondary Trunk Table

6. Contact Registrant Tables 

Contact Registrant Tables are used to manage contacts that are registered to a SIP server. The SIP Server Configuration can specify a Contact Registrant Table, and use the username portion of the table for outbound calls.

Select Settings > SIP > Contact Registrant Tables to access the Contact Registrant Tables configuration screen.

Figure : CenturyLink Primary Trunk Table

 

Figure : CenturyLink Secondary Trunk Table

7. Transformation

Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. For example, transformations can convert a public PSTN number into a private extension number, or into a SIP address (URI). Every entry in a Call Routing Table requires a Transformation Table, and are sequentially selected. In addition, Transformation Tables are configurable as a reusable pool that Action Sets can reference.

Select Settings > Transformation to access the Transformation configuration screen.

Figure : CenturyLink Transformation


Figure : CUCM 10.5 Transformation

8. Signaling Groups

Signaling Groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. These groups are the entity to which calls are routed, as well as the location from which Call Routes are selected. These are also the locations from which Tone Tables and Action Sets are selected. In the case of SIP, this will specify protocol settings and link to server, media and mapping tables.

Select Settings > Signaling Groups to access the Signaling Groups configuration screen.

Figure : CenturyLink Primary Signaling Group

Figure : CenturyLink Secondary Signaling Group


Figure : CUCM 10.5 Primary Signaling Group


Figure : CUCM 10.5 Secondary Signaling Group


Figure : Fax Signaling Group


9. Call Routing Table

Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports, and between protocols (like ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration of which calls will be carried, and also how the calls are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message translations, Cause Code Reroutes, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).

Select Settings > Call Routing Table to access the Call Routing Table configuration screens.

Figure : CenturyLink Primary Trunk Call Routing

Figure : CenturyLink Secondary Trunk Call Routing

                                                                                                                               

Figure : Fax Call Routing

10. Cause Code Reroute

Terminating any calls returns a Q.850 Cause Code when they end. These codes can be used to determine whether or not to reroute the call to another signalling group. A Cause Code Reroute table contains one or more Q.850 Cause Codes that, when matched, trigger a reroute.

Select Settings > Telephony Mapping Tables > Cause Code Reroute to access the Cause Code Reroute configuration screen.

Figure : Cause Code Reroute

11. Condition Rule Table

Condition rules are simple rules that apply to a specific component of a message (for example, diversion.uri.host, from.uri.host, etc.), and the value of the field specified in the Match Type list box is matched against a literal value, token, or REGEX.

Select Settings > SIP > Message Manipulation > Condition Rule Table to access the Condition Rule Table screen.

Figure : Condition Rule ID 6

Figure : Condition Rule ID 7

Figure : Condition Rule ID 8

 

11. Message Rule Table

The SIP Message Manipulation feature is used by a SIP Signaling Group to manipulate the incoming or outgoing messages. This feature is intended to enhance interoperability with different vendor equipment and applications, and for correcting any fixable protocol errors in SIP messages while in progress without any changes to firmware/software.

Select Settings > SIP > Message Manipulation > Message Rule Table to access the Message Rule Table screen.

Figure : CenturyLink Invite Message Rule 1

Figure : CenturyLink Invite Message Rule 2

Figure : CenturyLink Invite Message Rule 3

Figure : CenturyLink Invite Message Rule 4

Figure : CenturyLink Invite Message Rule 5

Figure : CenturyLink Invite Message Rule 6

Figure : CenturyLink Invite Message Rule 7

Figure : CenturyLink Invite Message Rule 8

Figure : CenturyLink Invite Message Rule 9


Figure : CenturyLink Register Message Rule 1

Figure : CenturyLink Register Message Rule 2


Figure : CenturyLink Register Message Rule 3


 

Test Results

 

Table : Test Results

External IDTitleDescriptionTest SetupStatus Comments
g729-001Anonymous Call Rejection ActivatePBX User dials *77
PSTN Calls PBX User with Caller ID Block
Should receive an announcement
*77 is Dialed PBX and leaves PBX Phones gets an announcement
Calling Party blocks caller ID
Calling party makes a call to PBX User Calling Party receives an announcement when PBX user is dialed
Passed 
g729-002Anonymous Call Rejection Deactivate PBX User dials *87
PSTN Calls PBX User with Caller ID block
Call Should Complete

*87 is dialed PBX User receives and announcement
PSTN calls PBX User
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye

Passed 
g729-003Anonymous Call PBX-BW PBX sends anonymous call to BW
BW delivers the calls Private or unknown or anonymous to PSTN
PBX is configured to send a call to BW as anonymous with TN as PSTN
BW delivers the call to PSTN as Private or Anonymous
PSTN phone shows the call as Private or Anonymous
Call is answered by PSTN
PBX user hangs up the call
Passed 
g729-004Alien TNs A call PBX call originate where the From TN is not part of the customer trunk group.  As long as the pilot number is identified in the outgoing call by PAI, the BroadWorks will accept and route the call.After Alien TN is set up on a Trunk in CenturyLink Network,
PBX User Places a Call to PSTN
PBX User receives ringback
PSTN receives ringing
PSTN receives caller id of the Alien TN
PSTN answers the call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Passed 
g729-005Barge In Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33 +PBX User Ext
PSTN, User 1, and User 2 should be conf
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1 Extension
PSTN, PBX User 1, and PBX User 2 conference together
2 Way Audio is heard by all Legs
PBX User 1 drops from Call
2 way Audio is heard by PSTN and PBX User 2
PSTN drops call
PBX User 2 receives a Bye
Passed 
g729-006Barge In Exempt In the Portal Enable Barge In Exempt
Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33 +PBX User Ext
User 2 Should not be conf
Barge in Exempt is set on PBX user 1
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1 Extension
PBX user 2 is not allowed to barge in
PSTN drops the call
PBX User 1 receives a Bye
Passed 
g729-007PSTN to BWAPSTN calls BWA Number
Enter Calling Number (2nd Phone Location)
Enter Called Number  (PSTN)
PSTN should Ring with Caller ID of 2nd Phone
Answer Call
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
Announcement is received
Enter calling Number (2nd Phone created in BWA)
Announcement received
Enter Called Number (PSTN 2)
PSTN 1 receives ringback
PSTN 2 receives ringing
PSTN 2 receives caller ID of 2nd Phone (Not of PSTN 1)
PSTN 2 Answers Call
2 way audio is received
PSTN 2 releases Calls
PSTN receives a Bye
BlockedAnywhere service is not activated for test account
g729-008PSTN to PBX user with BWAPSTN Calls User with BWA
PBX User and 2nd Location should Ring
Answer phone for 2nd location
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
PSTN 1 receives ringback
Both PBX User and 2nd Phone Location Number gets ringing
Both PBX User and 2nd Phone Location Number gets Caller ID of PSTN
Call is answered on Location 2
PBX User no longer gets ringing (cancel)
2 way Audio
Location 2 releases call
PSTN receives a Bye
BlockedAnywhere service is not activated for test account
g729-009Call Forwarding Always Activate PBX User dials *72
Enter the CFA Destination TN
PSTN calls PBX User with CFA
PBX User 1 Dials *72
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g729-010Call Forwarding Always InterrogatePBX User with CFA dials
*21*
Announcement received
PBX User 1 Dials *21*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-011Call Forwarding Always  Deactivate PBX User with CFA dials *73
PSTN Calls PBX User 
PBX User 1 Dials *73
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-012Call Forwarding Always to Voicemail Activate PBX User Dials *21
PSTN Dials PBX User with CFA
Verify Call goes to Voicemail
PBX User 1 Dials *21
Announcement is received
When announcement completes PBX User receives a Bye
PSTN User Calls PBX User 1
Call should go directly to voicemail
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g729-013Call Forwarding Always to Voicemail Deactivate PBX User with CFA dial #21
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #21
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-014PSTN call is CFB to PSTN with ID RestrictedPBX configured to send CFB to BW for identified Station.
BW is configured with CFB to PSTN2.
PSTN 1 Calls PBX  with Caller ID Restricted
PSTN 1 hears ring back
PBX send 486 Busy to BW
BW forwards the call to PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays Private/Anonymous
PSTN 2 Answers the call.
Two way voice path is established between PSTN 1 and PSTN 2
PSTN 2 hangs up
PSTN2 should receive Private/Anonymous as CLIDPassed 
g729-015PSTN with Privacy call to PBX is CFA to PSTN PBX User  is configured  with CFA to PSTN 2
PSTN 1 Calls PBX  with Caller ID Restricted
PSTN 1 hears ring back
PBX sends a new call to BW with PSTN 2 Number, From as Anonymous and PAI set to Pilot Number
BW forwards the call to PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays Pilot Number
PSTN 2 Answers the call.
Two way voice path is established between PSTN 1 and PSTN 2
PSTN 2 hangs up
Pilot Number should be shown as CLID on PSTN2Passed 
g729-016Call Forwarding Busy ActivatePBX User dials *90
Enter the CFB Destination TN
PSTN calls PBX User with CFB
PBX User 1 Dials *90
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Busy PBX User 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g729-017Call Forwarding Busy Interrogate PBX User with CFB dials
*67*
Announcement received
PBX User 1 Dials *67*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-018Call Forwarding Busy Deactivate PBX User with CFB dials *91
PSTN Calls PBX User 
PBX User 1 Dials *91
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-019Call Forwarding Busy to Voicemail ActivatePBX User Dials *40
PSTN Dials PBX User with CFB
Verify Call goes to Voicemail
PBX User 1 Dials *40
Announcement is received
When announcement completes PBX User receives a Bye
Busy PBX User 1
PSTN User Calls PBX User 1
Call should go directly to voicemail
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g729-020Call Forwarding Busy to Voicemail Deactivate PBX User with CFB dial #40
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #40
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-021Call Forwarding No Answer Activate PBX User dials *92
Enter the CFNA Destination TN
PSTN calls PBX User with CFNA
PBX User 1 Dials *92
Announcement is heard
PBX User enters PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g729-022Call Forwarding No Answer- RNA TimerPBX User dials *610
Enter 1 #
PSTN calls PBX User with CFNA
Verify Call is forwarded
PBX User 1 Dials *610
Announcement is Heard
PBX User enter 1 for amount of Rings
After announcement completes PBX User 1 receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
PassedMinimum is 0 or 2 rings which can be entered
g729-023Call Forwarding No Answer Interrogate PBX User with CFNA dials
*61*
Announcement received
PBX User 1 Dials *61*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-024Call Forwarding No Answer Deactivate PBX User with CFNA dials *93
PSTN Calls PBX User 
PBX User 1 Dials *93
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-025Call Forwarding No Answer to Voicemail ActivatePBX User Dials *41
PSTN Dials PBX User with CFNA
Verify Call goes to Voicemail
PBX User 1 Dials *41
Announcement is received
When announcement completes PBX User receives a Bye
Busy PBX User 1
PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g729-026Call Forwarding No Answer to Voicemail Deactivate PBX User with CFNA dial #41
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #41
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-027Call Forwarding Not Reachable Activate PBX User dials *94
Enter the CFNR Destination TN
Unregister Pilot TNs
PSTN calls PBX User with CFNR
Verify Call is forwarded
Register Pilot TNs
PBX User 1 Dials *94
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Unplug SBC Lan Cable
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PSTN User 2 gets ringing
PSTN user 2 receives Caller ID (PSTN Originator Caller)
PSTN User answers call
2 way Audio
PSTN User 1 releases call
PSTN User 2 receives a Bye
Passed 
g729-028Call Forwarding Not Reachable Interrogate PBX User with CFNR dials
*63*
Announcement received
PBX User 1 Dials *63*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-029Call Forwarding Not Reachable DeactivatePBX User with CFNR dials *95
PSTN Calls PBX User 
PBX User 1 Dials *95
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-030Call Forwarding Selective ActivateLog into Portal and set up Call forward selective User with a PSTN Number
PBX User with CFS enters #76
PSTN User calls PBX User with CFS
Call should be call forwarded
Log into Portal and set up Call forward selective User with a PSTN Number
PBX User with CFS dials #76
Announcement received
PBX User receives a Bye
From a Selected PSTN Dial PBX User 1
PBX User should not Ring
Call should be call forwarded to the CFS Destination
PSTN receives Ringback
Destination receives Ringing
Destination receives Caller ID (Originator PSTN)
Destination answers call
2 way Audio
PSTN ends the call
Destination receives a Bye
Passed 
g729-031Call Forwarding Selective DeactivatePBX User with CFS enters #77
PSTN User calls PBX User
Call should not be forwarded
PBX User 1 Dials #77
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g729-032Call Return by PBX User PBX User dials *69PSTN 1 Calls PBX User 1
PSTN 1 receives ringback
PBX User 1 receives ringing
PBX User 1 receives caller ID
PBX User 1 answers call
2 way Audio
PSTN 1 ends the call
PBX User 1 receives a Bye
PBX User 1 Dials *69
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN receives Caller ID
PSTN answers
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed 
g729-033Consultative Transfer with SIP REFER PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
PBX User has Audio with PSTNs
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PSTN2 now have audio
 Not SupportedCUCM 10.5 does not support outbound SIP Transfer with Refer method
g729-034Unattended Transfer with SIP REFER PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
During Ringback PBX User transfers
PSTN 1 has MOH
PSTN2 answers call
PSTN and PSTN2 now have audio
 Not SupportedCUCM 10.5 does not support outbound SIP Transfer with Refer method
g729-035Consultative Transfer  PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
PBX User 1 has Audio with PBX User 2
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PBX 2 now have audio

PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User 1
PBX User 2 answers the Call
2 way Audio
PBX User 1 transfers the call
MOH Ends
PSTN 1 and PBX User 2 are now connected
2 Way Audio
PSTN 1 Ends the call
PBX User 2 receives the Bye
Passed 
g729-036Unattended Transfer PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
During Ringback PBX User transfers
PSTN 1 has MOH
PBX User 2 answers call
PSTN and PBX User 2 now have audio

PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN 1
PBX User 1 release call
PBX User 2 answers the Call
MOH Ends
2 way Audio
PSTN 1 release the call
PBX User 2 receives the Bye
Passed 
g729-037Call Waiting Persistent ActivatePBX User dials *43
PSTN Calls PBX User
PSTN 2 Calls PBX User
Verify Call Waiting Tone
PBX User 1 Dials *43
Announcement is heard
PBX Receives a Bye
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User 2 Calls PBX User 1
PSTN User 2 receives ringback
PBX User 1 receives caller ID
PBX User 1 hear Call Waiting Tone
PBX User Places PSTN User 1 on Hold
PSTN User 1 hears MOH
PBX User 1 answers Call from PSTN 2
2 way Audio
Verify PBX User 1 can swap between to callers
While on PBX User 1 and PSTN User 1
PSTN 1 releases Call
PBX User 1 receives a Bye
Call 2 should still be up with PSTN 2 hearing MOH
Passed 
g729-038Call Waiting Persistent DeactivatePBX User Dials #43
PSTN Calls PBX User
PSTN 2 Calls PBX User
Call 2 should go to voicemail
PBX User 1 Dials #43
Announcement is heard
PBX User 1 Receives a Bye after Announcement is completed
PSTN User Calls PBX User 1
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 answers call
2 way Audio
PSTN User 1 releases the call
PBX User 1 receives a Bye
PassedCall Forwarding Busy to Voicemail  is activated to send PSTN User 2 to voicemail
g729-039Customer Originated Trace PSTN Calls PBX User
PBX User Answers the Call
PBX User Hangs up call
PBX User enters *57
Verify announcement
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
PBX User 1 Dial *57
Announcement received
Announcement Completes PBX User receives a Bye
Passed 
g729-040Enhanced Call Logs Log into portal and verify Call logsLog into the portal for PBX User 1
On main screen verify calls Logs are displayed
Missed
Received
Placed
Passed 
g729-041Last Number Redial PBX User dials *66
The last number dialed should be called
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
PBX User 1 Dial *66
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed 
g729-042MOH Verify MOH for conference, transfer, and holdPSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PBX User 1 Places call on Hold
PSTN receives MOH
PBX User retrieves call from Hold
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g729-043Remote Office  - Like CFAProvision Remote office for a SIP Trunk user on the BroadWorks portal to use PSTN number A. Place a call from a PSTN number B to the SIP Trunk user's DID and verify that it is forwarded to PSTN number A (the destination configured in BroadWorks). Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
PSTN User 2 Calls PBX User 1
PSTN 2 receives ringback
PSTN User 1  gets ringing with PSTN 2 Caller ID and Diversion header for PBX User1
PSTN User 1 answers call
2 way Audio
PSTN 1 releases call
Passed 
g729-044Remote Office - Quick CallProvision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal,  Manage Users, select Configure Features of User 1, under Quick Call, add PSTN B number and click on the Call Button. PSTN A should Start Ringing with PBX User 1 Caller ID.  Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Initiate a Quack Call to PSTN 2 on the portal
PSTN User 1  gets ringing with PBX User 1  Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed 
g729-045Remote Office - Click to  CallProvision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal,  Manage Users, select Configure Features of User 1, under Call Logs, select either incoming/outgoing/missed calls and  Click on a  Call under Phone Number Click To call column. PSTN A should Start Ringing with PBX User 1 Caller ID.  Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Review call logs and identify a call log that needs to be called via Click to Call.
Click on the identified call log under Click to Call
PSTN User 1  gets ringing with PBX User 1  Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed 
g729-046Selective Call Acceptance Provision selective call acceptance in the BroadWorks portal. Place a call from an accepted TN to the SIP Trunk User. Verify that the call completes normally. Place a call from a TN that is not on the accept list and verify that the call does not reach the  SBC.Log into the portal for PBX User 1
Set up Selected Call Acceptance to PSTN Number 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases Call
PBX User 1 receives a Bye
Passed 
g729-047Selective Call Rejection Provision selective call rejection in the BroadWorks portal. Place a call from a TN not on the reject list to the SIP Trunk User. Verify that the call completes normally. Place a call from a TN that is on the reject list and verify that the call does not reach the SBC.Log into the portal for PBX User 1
Set up Selected Call rejection to PSTN Number 1
PSTN Calls PBX User 1
Verify PSTN gets an announcement
PSTN receives a Bye
Passed 
g729-048Sequential RingProvision sequential ring in the BroadWorks portal. Place a call to the SIP trunk user. Verify that the numbers in the sequential ring list are dialed in order.Log into the Portal for PBX User 1
Set up Sequential Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer PBX User 1 receives a Cancel
PBX User 2 gets ringing
PBX user 2 receives Caller ID
After RNA Timer PBX User 1 receives a Cancel
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 answers call
2 way Audio
PSTN releases Call
PBX User 3 receives a Bye
Passed 
g729-049Simultaneous Ring Provision Simultaneous ring in the BroadWorks portal. Place a call to the SIP trunk user. Verify that the numbers in the Simultaneous ring list are dialed at once.Log into the Portal for PBX User 1
Set up Simultaneous Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 Answers Call
PBX User 1 and 2 receive a Cancel
2 way Audio
PSTN releases Call
PSTN User 3 receives a Bye
Passed 
g729-050Third Party MWI Control NOTIFY  Provision Third Party MWI in the BroadWorks portal. Provision the CT Voice Mail system to notify BroadWorks of unread messages in the user's voice mail box. Confirm that the NOTIFY is sent to BroadWorks and that the NOTIFY is sent to the PBX.PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Log into Mailbox
Listen To Voicemail
Delete Voicemail
Verify MWI is gone
PBX User 1 ends the Call
PassedCall Forwarding No Answer to Voicemail is activated to send PSTN user 1 to voicemail after timer
g729-051Voice Mail ConsultationProvision Voice Mail n the BroadWorks or NYMPH portal. Provision the PBX to forward calls to an external voice mail system as the user's call coverage. Confirm the PBX user's capability to retrieve voice mail from the external Voice Mail system.PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Log into Mailbox
Listen To Voicemail
Delete Voicemail
Verify MWI is gone
PBX User 1 ends the Call
Passed 
g729-052PBX Initiate ConferencePBX User Calls PSTN
PBX User Conferences PBX User 2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User conferences call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User 1
PBX User 2 answers the Call
2 way Audio
PBX User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PBX User 1 Ends the call
PBX User 2 and PSTN receives the Bye
Passed 
g729-053PSTN Initiate ConferencePBX User calls PSTN
PSTN conferences PBX User2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PSTN User 1 conferences call to PBX User 2
PBX User 1 gets MOH
PSTN User 1 gets Dial tone
PSTN User 1 dials PBX User 2 Extension
PSTN User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN User 1
PBX User 2 answers the Call
2 way Audio
PSTN User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PSTN User 1 Ends the call
PBX User 1 and PBX User 2 Still Have Audio
PBX User 1 End the Call
PBX User 2 receives a Bye
Passed 
g729-054Huntgroup Seq RingPSTN Calls Huntgroup Seq ring
Answer call on 2nd Member
Log into Admin Portal
Create Huntgroup with 3 members
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer is reached
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 Answers the call
2 way Audio
PSTN ends the call
PBX User 2 receives a Bye
BlockedHunt group is not purchased
g729-055Huntgroup Seq Ring RNA to VoicemailPSTN calls Huntgroup Seq ring
RNA to Voicemail
Log into Admin Portal
Create Huntgroup with 3 members
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer is reached
PBX User 2 gets ringing
PBX user 2 receives Caller ID
After RNA Timer is reached
PBX User 3 gets ringing
PBX user 3 receives Caller ID
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Enter *#
Log into HuntGroup Mailbox
Listen To Voicemail
Delete Voicemail
PBX User 1 ends the Call
BlockedHunt group is not purchased
g729-056Huntgroup Sim RingPSTN calls Huntgroup Sim ring 3 members
Answer Call
Log into Admin Portal
Create Huntgroup with 3 members with Sequential Ring
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX user 3 receives Caller ID
PBX User 3 Answers the call
PBX User 3 Answers the Call
2 way Audio
PSTN ends the call
PBX User 2 receives a Bye
BlockedHunt group is not purchased
g729-057PBX to PBXPBX User Calls PBX User2 Same Trunk
Verify RTP is dropped to SBC
PBX User 1 Calls PBX User 2
PBX  User 1 receives ringback
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio RTP is on SBC/PBX
PBX User 1 End the call
PBX User 2 receives a Bye
Passed 
g729-058PSTN to PBXPSTN to PBX UserPSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g729-059PBX to PSTNPBX User to PSTNPBX User 1 Calls PSTN User 1
PBX User 1 receives ringback
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g729-060PBX to PBX Different PBX (diff realm)PBX User to PBX User Different PBX (diff realm)PBX User 1 Calls PBX User 2 Diff Realm
PBX  User 1 receives ringback
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio RTP
PBX User 1 End the call
PBX User 2 receives a Bye
Passed 
g729-061PSTN to PBXPSTN to PBX User Fax CallPSTN User 1 Fax Calls PBX User 1 Fax
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 Fax answers call
Fax is received
PBX User Ends The Call
PSTN User 1 receives a Bye
Passed 
g729-062PBX to PSTNPBX User to PSTN Fax CallPBX User 1 Fax Calls PSTN User 1 Fax
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User 1 Fax answers call
Fax is received
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g729-063PSTN to PBX -T38PSTN to PBX User Fax CallPSTN User 1 Fax Calls PBX User 1 Fax
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 Fax answers call
Fax is received
PBX User Ends The Call
PSTN User 1 receives a Bye
Passed 
g729-064PBX to PSTN -T38PBX User to PSTN Fax CallPBX User 1 Fax Calls PSTN User 1 Fax
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User 1 Fax answers call
Fax is received
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g729-065PBX to PSTN - Packet Marking for SIG packetsPBX to PSTN Call to verify that signaling packets are marked with DSCP = 24 (0x18)All outgoing SIP Signaling packets are marked with DSCP=24PassedSame as g729-059
g729-066PBX to PSTN - Packet Marking for RTP packetsPBX to PSTN Call to verify that rtp packets are marked with DSCP = 40 (0x28)All outgoing SIP RTP packets are marked with DSCP=40PassedSame as g729-059
g729-067PBX to PSTN - Directory assistancePBX User Calls PBX 411 and speaks with directory assistantPBX User 1 dials 411
Call is delivered to Directory Assistant for enquiry
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
Passed 
g729-068PBX to PSTN - Toll FreePBX User Calls 800.366.8201 to test toll free numbersPBX User 1 dials 800.366.8201 (CTL Support)
Call is delivered to CenturyLink Support
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
Passed 
g729-069PBX to PSTN - 911PBX User Calls 911 to get emergency supportPBX User 1 dials xxx-xxx-xxxx (CTL Rep)
Call is delivered to CenturyLink Rep
PBX User makes conferences 911 operator
PBX User, CTL rep and 911 operator are conferenced
????
Conditional PassedSame routing as for g729-067
g729-070PBX to PSTN - InternationalPBX User Calls international numberInternational Call is successfully established and torn down.Passed 
g711-001Anonymous Call Rejection ActivatePBX User dials *77
PSTN Calls PBX User with Caller ID Block
Should receive an announcement
*77 is Dialed PBX and leaves PBX Phones gets an announcement
Calling Party blocks caller ID
Calling party makes a call to PBX User Calling Party receives an announcement when PBX user is dialed
Passed 
g711-002Anonymous Call Rejection Deactivate PBX User dials *87
PSTN Calls PBX User with Caller ID block
Call Should Complete
*87 is dialed PBX User receives and announcement
PSTN calls PBX User
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Passed 
g711-003Anonymous Call PBX-BW PBX sends anonymous call to BW
BW delivers the calls Private or unknown or anonymous to PSTN
PBX is configured to send a call to BW as anonymous with TN as PSTN
BW delivers the call to PSTN as Private or Anonymous
PSTN phone shows the call as Private or Anonymous
Call is answered by PSTN
PBX user hangs up the call
Passed 
g711-004Alien TNs A call PBX call originate where the from TN that is not part of the customer trunk group.  As long as the pilot number is identified in outgoing call by PAI, the BroadWorks will accept and route the call.After Alien TN is set up on a Trunk in CenturyLink Network
PBX User Places a Call to PSTN
PBX User receives ringback
PSTN receives ringing
PSTN receives caller id of the Alien TN
PSTN answers the call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Passed 
g711-005Barge In Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33 +PBX User Ext
PSTN, User 1, and User 2 should be conf
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1 Extension
PSTN, PBX User 1, and PBX User 2 are conferenced together
2 Way Audio is heard by all Legs
PBX User 1 drops from Call
2 way Audio is heard by PSTN and PBX User 2
PSTN drops call
PBX User 2 receives a Bye
Passed 
g711-006Barge In Exempt In the Portal Enable Barge In Exempt
Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33 +PBX User Ext
User 2 Should not be conf
Barge in Exempt is set on PBX user 1
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1 Extension
PBX user 2 is not allowed to barge in
PSTN drops the call
PBX User 1 receives a Bye
Passed 
g711-007PSTN to BWAPSTN calls BWA Number
Enter Calling Number (2nd Phone Location)
Enter Called Number  (PSTN)
PSTN should Ring with Caller ID of 2nd Phone
Answer Call
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
Announcement is received
Enter calling Number (2nd Phone created in BWA)
Announcement received
Enter Called Number (PSTN 2)
PSTN 1 receives ringback
PSTN 2 receives ringing
PSTN 2 receives caller ID of 2nd Phone (Not of PSTN 1)
PSTN 2 Answers Call
2 way audio is received
PSTN 2 releases Calls
PSTN receives a Bye
BlockedAnywhere service is not activated for test account
g711-008PSTN to PBX user with BWAPSTN Calls User with BWA
PBX User and 2nd Location should Ring
Answer phone for 2nd location
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
PSTN 1 receives ringback
Both PBX User and 2nd Phone Location Number gets ringing
Both PBX User and 2nd Phone Location Number gets Caller ID of PSTN
Call is answered on Location 2
PBX User no longer gets ringing (cancel)
2 way Audio
Location 2 releases call
PSTN receives a Bye
BlockedAnywhere service is not activated for test account
g711-009Call Forwarding Always Activate PBX User dials *72
Enter the CFA Destination TN
PSTN calls PBX User with CFA
PBX User 1 Dials *72
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g711-010Call Forwarding Always InterrogatePBX User with CFA dials
*21*
Announcement received
PBX User 1 Dials *21*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-011Call Forwarding Always  Deactivate PBX User with CFA dials *73
PSTN Calls PBX User 
PBX User 1 Dials *73
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-012Call Forwarding Always to Voicemail Activate PBX User Dials *21
PSTN Dials PBX User with CFA
Verify Call goes to Voicemail
PBX User 1 Dials *21
Announcement is received
When announcement completes PBX User receives a Bye
PSTN User Calls PBX User 1
Call should go directly to voicemail
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g711-013Call Forwarding Always to Voicemail Deactivate PBX User with CFA dial #21
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #21
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-014PSTN with Privacy call to PBX is CFA to PSTN PBX User  is configured  with CFA to PSTN 2
PSTN 1 Calls PBX  with Caller ID Restricted
PSTN 1 hears ring back
PBX sends a new call to BW with PSTN 2 Number, From as Anonymous and PAI set to Pilot Number
BW forwards the call to PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays Pilot Number
PSTN 2 Answers the call.
Two way voice path is established between PSTN 1 and PSTN 2
PSTN 2 hangs up
Pilot Number should be shown as CLID on PSTN2Passed 
g711-015PSTN call is CFB to PSTN with ID RestrictedPBX configured to send CFB to BW for identified Station.
BW is configured with CFB to PSTN2.
PSTN 1 Calls PBX  with Caller ID Restricted
PSTN 1 hears ring back
PBX send 486 Busy to BW
BW forwards the call to PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays Private/Anonymous
PSTN 2 Answers the call.
Two way voice path is established between PSTN 1 and PSTN 2
PSTN 2 hangs up
PSTN2 should receive Private/Anonymous as CLIDPassed 
g711-016Call Forwarding Busy ActivatePBX User dials *90
Enter the CFB Destination TN
PSTN calls PBX User with CFB
PBX User 1 Dials *90
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Busy PBX User 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g711-017Call Forwarding Busy Interrogate PBX User with CFB dials
*67*
Announcement received
PBX User 1 Dials *67*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-018Call Forwarding Busy Deactivate PBX User with CFB dials *91
PSTN Calls PBX User 
PBX User 1 Dials *91
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-019Call Forwarding Busy to Voicemail ActivatePBX User Dials *40
PSTN Dials PBX User with CFB
Verify Call goes to Voicemail
PBX User 1 Dials *40
Announcement is received
When announcement completes PBX User receives a Bye
Busy PBX User 1
PSTN User Calls PBX User 1
Call should go directly to voicemail
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g711-020Call Forwarding Busy to Voicemail Deactivate PBX User with CFB dial #40
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #40
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-021Call Forwarding No Answer Activate PBX User dials *92
Enter the CFNA Destination TN
PSTN calls PBX User with CFNA
PBX User 1 Dials *92
Announcement is heard
PBX User enters PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g711-022Call Forwarding No Answer- RNA TimerPBX User dials *610
Enter 1 #
PSTN calls PBX User with CFNA
Verify Call is forwarded
PBX User 1 Dials *610
Announcement is Heard
PBX User enter 1 for amount of Rings
After announcement completes PBX User 1 receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
PassedMinimum is 0 or 2 rings which can be entered
g711-023Call Forwarding No Answer Interrogate PBX User with CFNA dials
*61*
Announcement received
PBX User 1 Dials *61*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-024Call Forwarding No Answer Deactivate PBX User with CFNA dials *93
PSTN Calls PBX User 
PBX User 1 Dials *93
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-025Call Forwarding No Answer to Voicemail ActivatePBX User Dials *41
PSTN Dials PBX User with CFNA
Verify Call goes to Voicemail
PBX User 1 Dials *41
Announcement is received
When announcement completes PBX User receives a Bye
Busy PBX User 1
PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g711-026Call Forwarding No Answer to Voicemail Deactivate PBX User with CFNA dial #41
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #41
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-027Call Forwarding Not Reachable Activate PBX User dials *94
Enter the CFNR Destination TN
Unregister Pilot TNs
PSTN calls PBX User with CFNR
Verify Call is forwarded
Register Pilot TNs
PBX User 1 Dials *94
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Unplug SBC Lan Cable
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PSTN User 2 gets ringing
PSTN user 2 receives Caller ID (PSTN Originator Caller)
PSTN User answers call
2 way Audio
PSTN User 1 releases call
PSTN User 2 receives a Bye
Passed 
g711-028Call Forwarding Not Reachable Interrogate PBX User with CFNR dials
*63*
Announcement received
PBX User 1 Dials *63*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-029Call Forwarding Not Reachable DeactivatePBX User with CFNR dials *95
PSTN Calls PBX User 
PBX User 1 Dials *95
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-030Call Forwarding Selective ActivateLog into Portal and set up Call forward selective User with a PSTN Number
PBX User with CFS enters #76
PSTN User calls PBX User with CFS
Call should be call forwarded
Log into Portal and set up Call forward selective User with a PSTN Number
PBX User with CFS dials #76
Announcement received
PBX User receives a Bye
From a Selected PSTN Dial PBX User 1
PBX User should not Ring
Call should be call forwarded to the CFS Destination
PSTN receives Ringback
Destination receives Ringing
Destination receives Caller ID (Originator PSTN)
Destination answers call
2 way Audio
PSTN ends the call
Destination receives a Bye
Passed 
g711-031Call Forwarding Selective DeactivatePBX User with CFS enters #77
PSTN User calls PBX User
Call should not be forwarded
PBX User 1 Dials #77
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g711-032Call Return by PBX User PBX User dials *69PSTN 1 Calls PBX User 1
PSTN 1 receives ringback
PBX User 1 receives ringing
PBX User 1 receives caller ID
PBX User 1 answers call
2 way Audio
PSTN 1 ends the call
PBX User 1 receives a Bye
PBX User 1 Dials *69
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN receives Caller ID
PSTN answers
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed 
g711-033Consultative Transfer with SIP REFER PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
PBX User has Audio with PSTNs
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PSTN2 now have audio
 Not SupportedCUCM 10.5 does not support outbound SIP Transfer with Refer method
g711-034Unattended Transfer with SIP REFER PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
During Ringback PBX User transfers
PSTN 1 has MOH
PSTN2 answers call
PSTN and PSTN2 now have audio
 Not SupportedCUCM 10.5 does not support outbound SIP Transfer with Refer method
g711-035Consultative Transfer  PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
PBX User 1 has Audio with PBX User 2
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PBX 2 now have audio

PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User 1
PBX User 2 answers the Call
2 way Audio
PBX User 1 transfers the call
MOH Ends
PSTN 1 and PBX User 2 are now connected
2 Way Audio
PSTN 1 Ends the call
PBX User 2 receives the Bye
Passed 
g711-036Unattended Transfer PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
During Ringback PBX User transfers
PSTN 1 has MOH
PBX User 2 answers call
PSTN and PBX User 2 now have audio

PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN 1
PBX User 1 release call
PBX User 2 answers the Call
MOH Ends
2 way Audio
PSTN 1 release the call
PBX User 2 receives the Bye
Passed 
g711-037Call Waiting Persistent ActivatePBX User dials *43
PSTN Calls PBX User
PSTN 2 Calls PBX User
Verify Call Waiting Tone
PBX User 1 Dials *43
Announcement is heard
PBX Receives a Bye
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User 2 Calls PBX User 1
PSTN User 2 receives ringback
PBX User 1 receives caller ID
PBX User 1 hear Call Waiting Tone
PBX User Places PSTN User 1 on Hold
PSTN User 1 hears MOH
PBX User 1 answers Call from PSTN 2
2 way Audio
Verify PBX User 1 can swap between to callers
While on PBX User 1 and PSTN User 1
PSTN 1 releases Call
PBX User 1 receives a Bye
Call 2 should still be up with PSTN 2 hearing MOH
Passed 
g711-038Call Waiting Persistent DeactivatePBX User Dials #43
PSTN Calls PBX User
PSTN 2 Calls PBX User
Call 2 should go to voicemail
PBX User 1 Dials #43
Announcement is heard
PBX User 1 Receives a Bye after Announcement is completed
PSTN User Calls PBX User 1
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 answers call
2 way Audio
PSTN User 1 releases the call
PBX User 1 receives a Bye
PassedCall Forwarding Busy to Voicemail  is activated to send PSTN User 2 to voicemail
g711-039Customer Originated Trace PSTN Calls PBX User
PBX User Answers the Call
PBX User Hangs up call
PBX User enters *57
Verify announcement
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
PBX User 1 Dial *57
Announcement received
Announcement Completes PBX User receives a Bye
Passed 
g711-040Enhanced Call Logs Log into portal and verify Call logsLog into the portal for PBX User 1
On main screen verify calls Logs are displayed
Missed
Received
Placed
Passed 
g711-041Last Number Redial PBX User dials *66
The last number dialed should be called
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
PBX User 1 Dial *66
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed 
g711-042MOH Verify MOH for conference, transfer, and holdPSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PBX User 1 Places call on Hold
PSTN receives MOH
PBX User retrieves call from Hold
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g711-043Remote Office  - Like CFAProvision Remote office for a SIP Trunk user on the BroadWorks portal to use PSTN number A. Place a call from a PSTN number B to the SIP Trunk user's DID and verify that it is forwarded to PSTN number A (the destination configured in BroadWorks). Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
PSTN User 2 Calls PBX User 1
PSTN 2 receives ringback
PSTN User 1  gets ringing with PSTN 2 Caller ID and Diversion header for PBX User1
PSTN User 1 answers call
2 way Audio
PSTN 1 releases call
Passed 
g711-044Remote Office - Quick CallProvision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal,  Manage Users, select Configure Features of User 1, under Quick Call, add PSTN B number and click on the Call Button. PSTN A should Start Ringing with PBX User 1 Caller ID.  Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Initiate a Quick Call to PSTN 2 on the portal
PSTN User 1  gets ringing with PBX User 1  Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed 
g711-045Remote Office - Click to  CallProvision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal,  Manage Users, select Configure Features of User 1, under Call Logs, select either incoming/outgoing/missed calls and  Click on a  Call under Phone Number Click To call column. PSTN A should Start Ringing with PBX User 1 Caller ID.  Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Review call logs and identify a call log that needs to be called via Click to Call.
Click on the identified call log under Click to Call
PSTN User 1  gets ringing with PBX User 1  Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed 
g711-046Selective Call Acceptance Provision selective call acceptance in the BroadWorks portal. Place a call from an accepted TN to the SIP Trunk User. Verify that the call completes normally. Place a call from a TN that is not on the accept list and verify that the call does not reach the  SBC.Log into the portal for PBX User 1
Set up Selected Call Acceptance to PSTN Number 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases Call
PBX User 1 receives a Bye
Passed 
g711-047Selective Call Rejection Provision selective call rejection in the BroadWorks portal. Place a call from a TN not on the reject list to the SIP Trunk User. Verify that the call completes normally. Place a call from a TN that is on the reject list and verify that the call does not reach the SBC.Log into the portal for PBX User 1
Set up Selected Call rejection to PSTN Number 1
PSTN Calls PBX User 1
Verify PSTN gets an announcement
PSTN receives a Bye
Passed 
g711-048Sequential RingProvision sequential ring in the BroadWorks portal. Place a call to the SIP trunk user. Verify that the numbers in the sequential ring list are dialed in order.Log into the Portal for PBX User 1
Set up Sequential Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer PBX User 1 receives a Cancel
PBX User 2 gets ringing
PBX user 2 receives Caller ID
After RNA Timer PBX User 1 receives a Cancel
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 answers call
2 way Audio
PSTN releases Call
PBX User 3 receives a Bye
Passed 
g711-049Simultaneous Ring Provision Simultaneous ring in the BroadWorks portal. Place a call to the SIP trunk user. Verify that the numbers in the Simultaneous ring list are dialed at once.Log into the Portal for PBX User 1
Set up Simultaneous Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 Answers Call
PBX User 1 and 2 receive a Cancel
2 way Audio
PSTN releases Call
PSTN User 3 receives a Bye
Passed 
g711-050Third Party MWI Control NOTIFY  Provision Third Party MWI in the BroadWorks portal. Provision the CT Voice Mail system to notify BroadWorks of unread messages in the user's voice mail box. Confirm that the NOTIFY is sent to BroadWorks and that the NOTIFY is sent to the PBX.PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Log into Mailbox
Listen To Voicemail
Delete Voicemail
Verify MWI is gone
PBX User 1 ends the Call
Passed 
g711-051Voice Mail ConsultationProvision Voice Mail n the BroadWorks or NYMPH portal. Provision the PBX to forward calls to an external voice mail system as the user's call coverage. Confirm the PBX user's capability to retrieve voice mail from the external Voice Mail system.PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Log into Mailbox
Listen To Voicemail
Delete Voicemail
Verify MWI is gone
PBX User 1 ends the Call
Passed 
g711-052PBX Initiate ConferencePBX User Calls PSTN
PBX User Conferences PBX User 2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User conferences call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User 1
PBX User 2 answers the Call
2 way Audio
PBX User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PBX User 1 Ends the call
PBX User 2 and PSTN receives the Bye
PassedPBX user 2 and PSTN still have an audio on after PBX user 1 ends the call
g711-053PSTN Initiate ConferencePBX User calls PSTN
PSTN conferences PBX User2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PSTN User 1 conferences call to PBX User 2
PBX User 1 gets MOH
PSTN User 1 gets Dial tone
PSTN User 1 dials PBX User 2 Extension
PSTN User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN User 1
PBX User 2 answers the Call
2 way Audio
PSTN User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PSTN User 1 Ends the call
PBX User 1 and PBX User 2 Still Have Audio
PBX User 1 End the Call
PBX User 2 receives a Bye
Passed 
g711-054Huntgroup Seq RingPSTN Calls Huntgroup Seq ring
Answer call on 2nd Member
Log into Admin Portal
Create Huntgroup with 3 members
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer is reached
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 Answers the call
2 way Audio
PSTN ends the call
PBX User 2 receives a Bye
BlockedHunt group is not purchased
g711-055Huntgroup Seq Ring RNA to VoicemailPSTN calls Huntgroup Seq ring
RNA to Voicemail
Log into Admin Portal
Create Huntgroup with 3 members
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer is reached
PBX User 2 gets ringing
PBX user 2 receives Caller ID
After RNA Timer is reached
PBX User 3 gets ringing
PBX user 3 receives Caller ID
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Enter *#
Log into HuntGroup Mailbox
Listen To Voicemail
Delete Voicemail
PBX User 1 ends the Call
BlockedHunt group is not purchased.
g711-056Huntgroup Sim RingPSTN calls Huntgroup Sim ring 3 members
Answer Call
Log into Admin Portal
Create Huntgroup with 3 members with Sequential Ring
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX user 3 receives Caller ID
PBX User 3 Answers the call
PBX User 3 Answers the Call
2 way Audio
PSTN ends the call
PBX User 2 receives a Bye
BlockedHunt group is not purchased.
g711-057PBX to PBXPBX User Calls PBX User2 Same Trunk
Verify RTP is dropped to SBC
PBX User 1 Calls PBX User 2
PBX  User 1 receives ringback
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio RTP is on SBC/PBX
PBX User 1 End the call
PBX User 2 receives a Bye
Passed 
g711-058PSTN to PBXPSTN to PBX UserPSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g711-059PBX to PSTNPBX User to PSTNPBX User 1 Calls PSTN User 1
PBX User 1 receives ringback
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g711-060PBX to PBX Different PBX (diff realm)PBX User to PBX User Different PBX (diff realm)PBX User 1 Calls PBX User 2 Diff Realm
PBX  User 1 receives ringback
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio RTP
PBX User 1 End the call
PBX User 2 receives a Bye
Passed 
g711-061PSTN to PBX -PassthroughPSTN to PBX User Fax CallPSTN User 1 Fax Calls PBX User 1 Fax
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 Fax answers call
Fax is received
PBX User Ends The Call
PSTN User 1 receives a Bye
Passed 
g711-062PBX to PSTN -PassthroughPBX User to PSTN Fax CallPBX User 1 Fax Calls PSTN User 1 Fax
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User 1 Fax answers call
Fax is received
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g711-063PSTN to PBX -T38PSTN to PBX User Fax CallPSTN User 1 Fax Calls PBX User 1 Fax
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 Fax answers call
Fax is received
PBX User Ends The Call
PSTN User 1 receives a Bye
Passed 
g711-064PBX to PSTN -T38PBX User to PSTN Fax CallPBX User 1 Fax Calls PSTN User 1 Fax
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User 1 Fax answers call
Fax is received
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g711-065PBX to PSTN - Packet Marking for SIG packetsPBX to PSTN Call to verify that signaling packets are marked with DSCP = 24 (0x18)All outgoing SIP Signaling packets are marked with DSCP=24PassedSame as g711-059
g711-066PBX to PSTN - Packet Marking for RTP packetsPBX to PSTN Call to verify that rtp packets are marked with DSCP = 40 (0x28)All outgoing SIP RTP packets are marked with DSCP=40PassedSame as g711-059
g711-067PBX to PSTN - Directory assistancePBX User Calls PBX 411 and speaks with directory assistantPBX User 1 dials 411
Call is delivered to Directory Assistant for enquiry
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
Passed 
g711-068PBX to PSTN - Toll FreePBX User Calls 800.366.8201 to test toll free numbersPBX User 1 dials 800.366.8201 (CTL Support)
Call is delivered to CenturyLink Support
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
Passed 
g711-069PBX to PSTN - 911PBX User Calls 911 to get emergency supportPBX User 1 dials xxx-xxx-xxxx (CTL Rep)
Call is delivered to CenturyLink Rep
PBX User makes conferences 911 operator
PBX User, CTL rep and 911 operator are conferenced
????
Conditional PassedSame routing as for g711-067
g711-070PBX to PSTN - InternationalPBX User Calls international numberInternational Call is successfully established and torn down.Passed 
External IDTitleDescriptionTest SetupStatusCommnets
 g729-001Configure Dual Trunk on PBXPBX is configured and connected to 2 PSTN GW/SBCsThe steps  will be  based on the type of PBX being utilized.
Ensure that trunks are configured between PBX and SBC.
Verify OPTIONS msgs from either PBX or SBC are being responded correctly by the other entity
Passed 
g729-002Configure Dual Trunk on ITSPITSP is configured and connected to 2 PSTN GW/SBCsThe steps  will be  based on the type of SBC being utilized.
Ensure the TWO SBCs are configured with individual trunks to ITSP
Passed 
g729-003Regitration of Dual TrunksEnsure that both trunks to ITSP are registered successfully using the individual trunk registration information1. Each SBC is configured with a trunk to ITSP and associated authentiation/digest and registration information.
2. Invoke a command on SBC to register the trunk with ITSP.
3. Verify that 200 OK is received from ITSP for both the trunks.
Passed 
g729-004Inbound PSTN calls pick correct trunk to SBCVerify that PSTN to PBX inbound calls arrive on both the trunks when multiple calls are made1.    Dial an inbound call to the PBX.
2.    Verify ringing is heard by calling and called parties.
3.    Verify the trace shows a valid ringing indication message
4.    Take called party phone off-hook.
5.    Verify that a media path is established in both directions.
6.    Hang up calling party
7.    Verify the IP/PBX receives a Bye message.
8.    Make a note of the Trunk on which the call arrived to the SBC and PBX.
9.    Repeat the above steps 3 more times (total 4 calls).
10.   Verify that calls to PBX arrive on both the trunks.
11.   Document Test Results.
12.   Save Trace.
Passed 
g729-005PBX calls are delivered to PSTN on both the trunksCalls from PBX to PSTN are delivered to ITSP/PSTN utilizing both the configured trunks1.    Dial an outbound call from the PBX.
2.    Verify ringing is heard by calling and called parties.
3.    Verify the trace shows a valid ringing indication message
4.    Take called party phone off-hook.
5.    Verify that a media path established in both directions.
6.    Hang up Calling Party
7.    Verify the IP/PBX sends a Bye message.
8.    Make a note of the Trunk on which the call was sent to ITSP.
9.    Repeat the above steps 3 more times (total 4 calls).
10.   Verify that calls from PBX are sent out on both the trunks to ITSP.
11.   Verify each call has PAI sent per the trunk configuration
12.   Document Test Results.
13.   Save Trace.
Passed 
g729-006Alien TN calls on 1st trunkVerify calls are successful with Alien TNs on 1st trunk1. After Alien TN is set up on a Trunk1 in CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases Calls
9. PSTN receives a Bye
Passed 
g729-007Alien TN calls on 2nd  trunkVerify calls are successful with Alien TNs on 2nd  trunk1. After Alien TN is set up on a Trunk2 in CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases Calls
9. PSTN receives a Bye
Passed 
g729-008Failover of 1st trunk WAN - PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-009Failover of 1st trunk WAN - PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-010Restore 1st trunk  WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has has been restored1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-011Restore 1st trunk  WAN: PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has has been restored1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-012Failover of 2nd trunk WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-013Failover of 2nd trunk WAN: PBX-PSTNEnsure that calls are delivered from PBX-PSTN when the second trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-014Restore 2nd trunk WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has has been restored1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-015Restore 2nd trunk WAN: PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the second trunk has has been restored1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-016Failover of 1st trunk LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN  when the first trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-017Failover of 1st trunk LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX  when the first trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-018Restore 1st trunk  LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has has been restored1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-019Restore 1st trunk  LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has has been restored1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-020Failover of 2nd trunk LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN  when the second trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-021Failover of 2nd trunk LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX  when the second trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-022Restore 2nd trunk  LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN when the second trunk has has been restored1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-023Restore 2nd trunk  LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has has been restored1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-001Configure Dual Trunk on PBXPBX is configured and connected to 2 PSTN GW/SBCsThe steps  will be  based on the type of PBX being utilized.
Ensure that trunks are configured between PBX and SBC.
Verify OPTIONS msgs from either PBX or SBC are being responded correctly by the other entity
Passed 
g711-002Configure Dual Trunk on ITSPITSP is configured and connected to 2 PSTN GW/SBCsThe steps  will be  based on the type of SBC being utilized.
Ensure the TWO SBCs are configured with individual trunks to ITSP
Passed 
g711-003Regitration of Dual TrunksEnsure that both trunks to ITSP are registered successfully using the individual trunk registration information1. Each SBC is configured with a trunk to ITSP and associated authentiation/digest and registration information.
2. Invoke a command on SBC to register the trunk with ITSP.
3. Verify that 200 OK is received from ITSP for both the trunks.
Passed 
g711-004Inbound PSTN calls pick correct trunk to SBCVerify that PSTN to PBX inbound calls arrive on both the trunks when multiple calls are made1.    Dial an inbound call to the PBX.
2.    Verify ringing is heard by calling and called parties.
3.    Verify the trace shows a valid ringing indication message
4.    Take called party phone off-hook.
5.    Verify that a media path is established in both directions.
6.    Hang up calling party
7.    Verify the IP/PBX receives a Bye message.
8.    Make a note of the Trunk on which the call arrived to the SBC and PBX.
9.    Repeat the above steps 3 more times (total 4 calls).
10.   Verify that calls to PBX arrive on both the trunks.
11.   Document Test Results.
12.   Save Trace.
Passed 
g711-005PBX calls are delivered to PSTN on both the trunksCalls from PBX to PSTN are delivered to ITSP/PSTN utilizing both the configured trunks1.    Dial an outbound call from the PBX.
2.    Verify ringing is heard by calling and called parties.
3.    Verify the trace shows a valid ringing indication message
4.    Take called party phone off-hook.
5.    Verify that a media path established in both directions.
6.    Hang up Calling Party
7.    Verify the IP/PBX sends a Bye message.
8.    Make a note of the Trunk on which the call was sent to ITSP.
9.    Repeat the above steps 3 more times (total 4 calls).
10.   Verify that calls from PBX are sent out on both the trunks to ITSP.
11.   Verify each call has PAI sent per the trunk configuration
12.   Document Test Results.
13.   Save Trace.
Passed 
g711-006Alien TN calls on 1st trunkVerify calls are successful with Alien TNs on 1st trunk1. After Alien TN is set up on a Trunk1 in CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases Calls
9. PSTN receives a Bye
Passed 
g711-007Alien TN calls on 2nd  trunkVerify calls are successful with Alien TNs on 2nd  trunk1. After Alien TN is set up on a Trunk2 in CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases Calls
9. PSTN receives a Bye
Passed 
g711-008Failover of 1st trunk WAN - PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-009Failover of 1st trunk WAN - PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-010Restore 1st trunk  WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has has been restored1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-011Restore 1st trunk  WAN: PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has has been restored1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-012Failover of 2nd trunk WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-013Failover of 2nd trunk WAN: PBX-PSTNEnsure that calls are delivered from PBX-PSTN when the second trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-014Restore 2nd trunk WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has has been restored1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-015Restore 2nd trunk WAN: PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the second trunk has has been restored1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-016Failover of 1st trunk LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN  when the first trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-017Failover of 1st trunk LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX  when the first trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-018Restore 1st trunk  LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has has been restored1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-019Restore 1st trunk  LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has has been restored1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-020Failover of 2nd trunk LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN  when the second trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-021Failover of 2nd trunk LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX  when the second trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-022Restore 2nd trunk  LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN when the second trunk has has been restored1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-023Restore 2nd trunk  LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has has been restored1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
 

 

Conclusion

This Application Notes document describes the configuration steps required for Sonus SBC 1000/2000 series to successfully interoperate with Cisco Unified Communications Manager 10.5 (CUCM 10.5). All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Test Results.