Table of Contents
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.
- For additional information on CUCM 10.5, please visit http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-communications-manager-callmanager/datasheet-c78-731839.html
- For additional information on Sonus SBC 1000/2000 series, please visit http://sonus.net.
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 Networks | SBC 2000 | V5.0.3build407 |
| Tenor AF | P108-09-21 | |
| Third-Party Equipment | Cisco Unified Communications Manager | 10.5.1.11005-2 |
| Cisco IP Phone 7942 | 9.3.1.57 |
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.
- Phone: (978) 614-8589 or (888) 391-3434 (Toll-free)
- Web: http://www.sonus.net/company/maintenance/log-trouble-tickets
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
Select System > Security > SIP Trunk Security Profile
Security Profiles
2.SIP Profile
Select Device > Device Settings > SIP Profile
SIP Profile
3.SIP Trunk
Select Device > Trunk > Add New
Primary SIP Trunk
Secondary SIP Trunk
4.Route Group
Select Call Routing > Route/Hunt > Route Group > Add New
Route Group
5.Route List
Select Call Routing > Route/Hunt > Route List > Add New
Route List
6.Route Pattern
Select Call Routing > Route/Hunt > Route Pattern > Add New
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:
- SIP Profiles
- SIP Server
- Media Profiles
- Media List
- Remote Authorization Tables
- Contact Registrant Tables
- Transformation
- Signaling Groups
- Call Routing Table
- Cause Code Reroute
- Condition Rule Table
- 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:
CenturyLink SIP Profile
CUCM 10.5 SIP Profile
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:
CenturyLink Primary SIP Server
CenturyLink Secondary SIP Server
CUCM 10.5 Primary SIP Server
CUCM 10.5 Secondary SIP Server
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.
CenturyLink Codecs
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.
CenturyLink Media List
CUCM 10.5 Media List
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 Creating and Modifying Entries in the SIP Servers (search online SBC 1000/2000 documentation).
Select Settings > SIP > Remote Authorization Tables to access the Remote Authorization Tables configuration screen.
CenturyLink Primary Trunk Table
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.
CenturyLink Primary Trunk Table
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.
CenturyLink Transformation
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.
CenturyLink Primary Signaling Group
CenturyLink Secondary Signaling Group
CUCM 10.5 Primary Signaling Group
CUCM 10.5 Secondary Signaling Group
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.
CenturyLink Primary Trunk Call Routing
CenturyLink Secondary Trunk Call Routing
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.
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.
Condition Rule ID 6
Condition Rule ID 7
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.
CenturyLink Invite Message Rule 1
CenturyLink Invite Message Rule 2
CenturyLink Invite Message Rule 3
CenturyLink Invite Message Rule 4
CenturyLink Invite Message Rule 5
CenturyLink Invite Message Rule 6
CenturyLink Invite Message Rule 7
CenturyLink Invite Message Rule 8
CenturyLink Invite Message Rule 9
CenturyLink Register Message Rule 1
CenturyLink Register Message Rule 2
CenturyLink Register Message Rule 3
Test Results
Test Results
| External ID | Title | Description | Test Setup | Status | Comments |
|---|---|---|---|---|---|
| g729-001 | Anonymous Call Rejection Activate | PBX 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 | |
| g729-002 | Anonymous 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 | Passed | |
| g729-003 | Anonymous 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 | |
| g729-004 | Alien 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 | |
| g729-005 | Barge 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 | |
| g729-006 | Barge 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 | |
| g729-007 | PSTN to BWA | PSTN 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 | Blocked | Anywhere service is not activated for test account |
| g729-008 | PSTN to PBX user with BWA | PSTN 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 | Blocked | Anywhere service is not activated for test account |
| g729-009 | Call 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 | |
| g729-010 | Call Forwarding Always Interrogate | PBX 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 | |
| g729-011 | Call 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 | |
| g729-012 | Call 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 | |
| g729-013 | Call 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 | |
| g729-014 | PSTN call is CFB to PSTN with ID Restricted | PBX 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 CLID | Passed | |
| g729-015 | PSTN 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 PSTN2 | Passed | |
| g729-016 | Call Forwarding Busy Activate | PBX 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 | |
| g729-017 | Call 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 | |
| g729-018 | Call 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 | |
| g729-019 | Call Forwarding Busy to Voicemail Activate | PBX 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 | |
| g729-020 | Call 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 | |
| g729-021 | Call 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 | |
| g729-022 | Call Forwarding No Answer- RNA Timer | PBX 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 | Passed | Minimum is 0 or 2 rings which can be entered |
| g729-023 | Call 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 | |
| g729-024 | Call 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 | |
| g729-025 | Call Forwarding No Answer to Voicemail Activate | PBX 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 | |
| g729-026 | Call 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 | |
| g729-027 | Call 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 | |
| g729-028 | Call 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 | |
| g729-029 | Call Forwarding Not Reachable Deactivate | PBX 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 | |
| g729-030 | Call Forwarding Selective Activate | Log 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 | |
| g729-031 | Call Forwarding Selective Deactivate | PBX 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 | |
| g729-032 | Call Return by PBX User | PBX User dials *69 | PSTN 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 | |
| g729-033 | Consultative 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 Supported | CUCM 10.5 does not support outbound SIP Transfer with Refer method | |
| g729-034 | Unattended 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 Supported | CUCM 10.5 does not support outbound SIP Transfer with Refer method | |
| g729-035 | Consultative 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 | |
| g729-036 | Unattended 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 | |
| g729-037 | Call Waiting Persistent Activate | PBX 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 | |
| g729-038 | Call Waiting Persistent Deactivate | PBX 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 | Passed | Call Forwarding Busy to Voicemail is activated to send PSTN User 2 to voicemail |
| g729-039 | Customer 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 | |
| g729-040 | Enhanced Call Logs | Log into portal and verify Call logs | Log into the portal for PBX User 1 On main screen verify calls Logs are displayed Missed Received Placed | Passed | |
| g729-041 | Last 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 | |
| g729-042 | MOH | Verify MOH for conference, transfer, and hold | 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 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 | |
| g729-043 | Remote Office - Like CFA | Provision 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 | |
| g729-044 | Remote Office - Quick Call | Provision 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 | |
| g729-045 | Remote Office - Click to Call | Provision 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 | |
| g729-046 | Selective 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 | |
| g729-047 | Selective 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 | |
| g729-048 | Sequential Ring | Provision 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 | |
| g729-049 | Simultaneous 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 | |
| g729-050 | Third 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 | Call Forwarding No Answer to Voicemail is activated to send PSTN user 1 to voicemail after timer |
| g729-051 | Voice Mail Consultation | Provision 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 | |
| g729-052 | PBX Initiate Conference | PBX 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 | |
| g729-053 | PSTN Initiate Conference | PBX 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 | |
| g729-054 | Huntgroup Seq Ring | PSTN 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 | Blocked | Hunt group is not purchased |
| g729-055 | Huntgroup Seq Ring RNA to Voicemail | PSTN 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 | Blocked | Hunt group is not purchased |
| g729-056 | Huntgroup Sim Ring | PSTN 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 | Blocked | Hunt group is not purchased |
| g729-057 | PBX to PBX | PBX 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 | |
| g729-058 | PSTN to PBX | PSTN to PBX User | 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 Ends The Call PBX User 1 receives a Bye | Passed | |
| g729-059 | PBX to PSTN | PBX User to PSTN | PBX 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 | |
| g729-060 | PBX 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 | |
| g729-061 | PSTN to PBX | PSTN to PBX User Fax Call | PSTN 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 | |
| g729-062 | PBX to PSTN | PBX User to PSTN Fax Call | PBX 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 | |
| g729-063 | PSTN to PBX -T38 | PSTN to PBX User Fax Call | PSTN 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 | |
| g729-064 | PBX to PSTN -T38 | PBX User to PSTN Fax Call | PBX 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 | |
| g729-065 | PBX to PSTN - Packet Marking for SIG packets | PBX to PSTN Call to verify that signaling packets are marked with DSCP = 24 (0x18) | All outgoing SIP Signaling packets are marked with DSCP=24 | Passed | Same as g729-059 |
| g729-066 | PBX to PSTN - Packet Marking for RTP packets | PBX to PSTN Call to verify that rtp packets are marked with DSCP = 40 (0x28) | All outgoing SIP RTP packets are marked with DSCP=40 | Passed | Same as g729-059 |
| g729-067 | PBX to PSTN - Directory assistance | PBX User Calls PBX 411 and speaks with directory assistant | PBX 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 | |
| g729-068 | PBX to PSTN - Toll Free | PBX User Calls 800.366.8201 to test toll free numbers | PBX 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 | |
| g729-069 | PBX to PSTN - 911 | PBX User Calls 911 to get emergency support | PBX 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 Passed | Same routing as for g729-067 |
| g729-070 | PBX to PSTN - International | PBX User Calls international number | International Call is successfully established and torn down. | Passed | |
| g711-001 | Anonymous Call Rejection Activate | PBX 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 | |
| g711-002 | Anonymous 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 | |
| g711-003 | Anonymous 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 | |
| g711-004 | Alien 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 | |
| g711-005 | Barge 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 | |
| g711-006 | Barge 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 | |
| g711-007 | PSTN to BWA | PSTN 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 | Blocked | Anywhere service is not activated for test account |
| g711-008 | PSTN to PBX user with BWA | PSTN 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 | Blocked | Anywhere service is not activated for test account |
| g711-009 | Call 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 | |
| g711-010 | Call Forwarding Always Interrogate | PBX 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 | |
| g711-011 | Call 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 | |
| g711-012 | Call 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 | |
| g711-013 | Call 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 | |
| g711-014 | PSTN 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 PSTN2 | Passed | |
| g711-015 | PSTN call is CFB to PSTN with ID Restricted | PBX 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 CLID | Passed | |
| g711-016 | Call Forwarding Busy Activate | PBX 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 | |
| g711-017 | Call 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 | |
| g711-018 | Call 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 | |
| g711-019 | Call Forwarding Busy to Voicemail Activate | PBX 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 | |
| g711-020 | Call 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 | |
| g711-021 | Call 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 | |
| g711-022 | Call Forwarding No Answer- RNA Timer | PBX 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 | Passed | Minimum is 0 or 2 rings which can be entered |
| g711-023 | Call 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 | |
| g711-024 | Call 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 | |
| g711-025 | Call Forwarding No Answer to Voicemail Activate | PBX 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 | |
| g711-026 | Call 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 | |
| g711-027 | Call 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 | |
| g711-028 | Call 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 | |
| g711-029 | Call Forwarding Not Reachable Deactivate | PBX 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 | |
| g711-030 | Call Forwarding Selective Activate | Log 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 | |
| g711-031 | Call Forwarding Selective Deactivate | PBX 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 | |
| g711-032 | Call Return by PBX User | PBX User dials *69 | PSTN 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 | |
| g711-033 | Consultative 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 Supported | CUCM 10.5 does not support outbound SIP Transfer with Refer method | |
| g711-034 | Unattended 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 Supported | CUCM 10.5 does not support outbound SIP Transfer with Refer method | |
| g711-035 | Consultative 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 | |
| g711-036 | Unattended 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 | |
| g711-037 | Call Waiting Persistent Activate | PBX 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 | |
| g711-038 | Call Waiting Persistent Deactivate | PBX 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 | Passed | Call Forwarding Busy to Voicemail is activated to send PSTN User 2 to voicemail |
| g711-039 | Customer 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 | |
| g711-040 | Enhanced Call Logs | Log into portal and verify Call logs | Log into the portal for PBX User 1 On main screen verify calls Logs are displayed Missed Received Placed | Passed | |
| g711-041 | Last 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 | |
| g711-042 | MOH | Verify MOH for conference, transfer, and hold | 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 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 | |
| g711-043 | Remote Office - Like CFA | Provision 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 | |
| g711-044 | Remote Office - Quick Call | Provision 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 | |
| g711-045 | Remote Office - Click to Call | Provision 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 | |
| g711-046 | Selective 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 | |
| g711-047 | Selective 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 | |
| g711-048 | Sequential Ring | Provision 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 | |
| g711-049 | Simultaneous 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 | |
| g711-050 | Third 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 | |
| g711-051 | Voice Mail Consultation | Provision 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 | |
| g711-052 | PBX Initiate Conference | PBX 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 | PBX user 2 and PSTN still have an audio on after PBX user 1 ends the call |
| g711-053 | PSTN Initiate Conference | PBX 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 | |
| g711-054 | Huntgroup Seq Ring | PSTN 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 | Blocked | Hunt group is not purchased |
| g711-055 | Huntgroup Seq Ring RNA to Voicemail | PSTN 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 | Blocked | Hunt group is not purchased. |
| g711-056 | Huntgroup Sim Ring | PSTN 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 | Blocked | Hunt group is not purchased. |
| g711-057 | PBX to PBX | PBX 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 | |
| g711-058 | PSTN to PBX | PSTN to PBX User | 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 Ends The Call PBX User 1 receives a Bye | Passed | |
| g711-059 | PBX to PSTN | PBX User to PSTN | PBX 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 | |
| g711-060 | PBX 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 | |
| g711-061 | PSTN to PBX -Passthrough | PSTN to PBX User Fax Call | PSTN 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 | |
| g711-062 | PBX to PSTN -Passthrough | PBX User to PSTN Fax Call | PBX 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 | |
| g711-063 | PSTN to PBX -T38 | PSTN to PBX User Fax Call | PSTN 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 | |
| g711-064 | PBX to PSTN -T38 | PBX User to PSTN Fax Call | PBX 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 | |
| g711-065 | PBX to PSTN - Packet Marking for SIG packets | PBX to PSTN Call to verify that signaling packets are marked with DSCP = 24 (0x18) | All outgoing SIP Signaling packets are marked with DSCP=24 | Passed | Same as g711-059 |
| g711-066 | PBX to PSTN - Packet Marking for RTP packets | PBX to PSTN Call to verify that rtp packets are marked with DSCP = 40 (0x28) | All outgoing SIP RTP packets are marked with DSCP=40 | Passed | Same as g711-059 |
| g711-067 | PBX to PSTN - Directory assistance | PBX User Calls PBX 411 and speaks with directory assistant | PBX 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 | |
| g711-068 | PBX to PSTN - Toll Free | PBX User Calls 800.366.8201 to test toll free numbers | PBX 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 | |
| g711-069 | PBX to PSTN - 911 | PBX User Calls 911 to get emergency support | PBX 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 Passed | Same routing as for g711-067 |
| g711-070 | PBX to PSTN - International | PBX User Calls international number | International Call is successfully established and torn down. | Passed | |
| External ID | Title | Description | Test Setup | Status | Commnets |
| g729-001 | Configure Dual Trunk on PBX | PBX is configured and connected to 2 PSTN GW/SBCs | The 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 | |
| g729-002 | Configure Dual Trunk on ITSP | ITSP is configured and connected to 2 PSTN GW/SBCs | The steps will be based on the type of SBC being utilized. Ensure the TWO SBCs are configured with individual trunks to ITSP | Passed | |
| g729-003 | Regitration of Dual Trunks | Ensure that both trunks to ITSP are registered successfully using the individual trunk registration information | 1. 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 | |
| g729-004 | Inbound PSTN calls pick correct trunk to SBC | Verify that PSTN to PBX inbound calls arrive on both the trunks when multiple calls are made | 1. 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 | |
| g729-005 | PBX calls are delivered to PSTN on both the trunks | Calls from PBX to PSTN are delivered to ITSP/PSTN utilizing both the configured trunks | 1. 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 | |
| g729-006 | Alien TN calls on 1st trunk | Verify calls are successful with Alien TNs on 1st trunk | 1. 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 | |
| g729-007 | Alien TN calls on 2nd trunk | Verify calls are successful with Alien TNs on 2nd trunk | 1. 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 | |
| g729-008 | Failover of 1st trunk WAN - PSTN-PBX | Ensure that calls are delivered from PSTN to PBX when the first trunk has failed on the WAN side | 1. 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 | |
| g729-009 | Failover of 1st trunk WAN - PBX-PSTN | Ensure that calls are delivered from PBX to PSTN when the first trunk has failed on the WAN side | 1. 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 | |
| g729-010 | Restore 1st trunk WAN: PSTN-PBX | Ensure that calls are delivered from PSTN to PBX when the first trunk has has been restored | 1. 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 | |
| g729-011 | Restore 1st trunk WAN: PBX-PSTN | Ensure that calls are delivered from PBX to PSTN when the first trunk has has been restored | 1. 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 | |
| g729-012 | Failover of 2nd trunk WAN: PSTN-PBX | Ensure that calls are delivered from PSTN to PBX when the second trunk has failed on the WAN side | 1. 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 | |
| g729-013 | Failover of 2nd trunk WAN: PBX-PSTN | Ensure that calls are delivered from PBX-PSTN when the second trunk has failed on the WAN side | 1. 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 | |
| g729-014 | Restore 2nd trunk WAN: PSTN-PBX | Ensure that calls are delivered from PSTN to PBX when the second trunk has has been restored | 1. 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 | |
| g729-015 | Restore 2nd trunk WAN: PBX-PSTN | Ensure that calls are delivered from PBX to PSTN when the second trunk has has been restored | 1. 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 | |
| g729-016 | Failover of 1st trunk LAN - PBX to PSTN | Ensure that calls are delivered from PBX to PSTN when the first trunk has failed on the LAN side | 1. 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 | |
| g729-017 | Failover of 1st trunk LAN - PSTN to PBX | Ensure that calls are delivered from PSTN to PBX when the first trunk has failed on the LAN side | 1. 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 | |
| g729-018 | Restore 1st trunk LAN - PBX to PSTN | Ensure that calls are delivered from PBX to PSTN when the first trunk has has been restored | 1. 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 | |
| g729-019 | Restore 1st trunk LAN - PSTN to PBX | Ensure that calls are delivered from PSTN to PBX when the first trunk has has been restored | 1. 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 | |
| g729-020 | Failover of 2nd trunk LAN - PBX to PSTN | Ensure that calls are delivered from PBX to PSTN when the second trunk has failed on the LAN side | 1. 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 | |
| g729-021 | Failover of 2nd trunk LAN - PSTN to PBX | Ensure that calls are delivered from PSTN to PBX when the second trunk has failed on the LAN side | 1. 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 | |
| g729-022 | Restore 2nd trunk LAN - PBX to PSTN | Ensure that calls are delivered from PBX to PSTN when the second trunk has has been restored | 1. 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 | |
| g729-023 | Restore 2nd trunk LAN - PSTN to PBX | Ensure that calls are delivered from PSTN to PBX when the second trunk has has been restored | 1. 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 | |
| g711-001 | Configure Dual Trunk on PBX | PBX is configured and connected to 2 PSTN GW/SBCs | The 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 | |
| g711-002 | Configure Dual Trunk on ITSP | ITSP is configured and connected to 2 PSTN GW/SBCs | The steps will be based on the type of SBC being utilized. Ensure the TWO SBCs are configured with individual trunks to ITSP | Passed | |
| g711-003 | Regitration of Dual Trunks | Ensure that both trunks to ITSP are registered successfully using the individual trunk registration information | 1. 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 | |
| g711-004 | Inbound PSTN calls pick correct trunk to SBC | Verify that PSTN to PBX inbound calls arrive on both the trunks when multiple calls are made | 1. 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 | |
| g711-005 | PBX calls are delivered to PSTN on both the trunks | Calls from PBX to PSTN are delivered to ITSP/PSTN utilizing both the configured trunks | 1. 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 | |
| g711-006 | Alien TN calls on 1st trunk | Verify calls are successful with Alien TNs on 1st trunk | 1. 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 | |
| g711-007 | Alien TN calls on 2nd trunk | Verify calls are successful with Alien TNs on 2nd trunk | 1. 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 | |
| g711-008 | Failover of 1st trunk WAN - PSTN-PBX | Ensure that calls are delivered from PSTN to PBX when the first trunk has failed on the WAN side | 1. 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 | |
| g711-009 | Failover of 1st trunk WAN - PBX-PSTN | Ensure that calls are delivered from PBX to PSTN when the first trunk has failed on the WAN side | 1. 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 | |
| g711-010 | Restore 1st trunk WAN: PSTN-PBX | Ensure that calls are delivered from PSTN to PBX when the first trunk has has been restored | 1. 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 | |
| g711-011 | Restore 1st trunk WAN: PBX-PSTN | Ensure that calls are delivered from PBX to PSTN when the first trunk has has been restored | 1. 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 | |
| g711-012 | Failover of 2nd trunk WAN: PSTN-PBX | Ensure that calls are delivered from PSTN to PBX when the second trunk has failed on the WAN side | 1. 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 | |
| g711-013 | Failover of 2nd trunk WAN: PBX-PSTN | Ensure that calls are delivered from PBX-PSTN when the second trunk has failed on the WAN side | 1. 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 | |
| g711-014 | Restore 2nd trunk WAN: PSTN-PBX | Ensure that calls are delivered from PSTN to PBX when the second trunk has has been restored | 1. 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 | |
| g711-015 | Restore 2nd trunk WAN: PBX-PSTN | Ensure that calls are delivered from PBX to PSTN when the second trunk has has been restored | 1. 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 | |
| g711-016 | Failover of 1st trunk LAN - PBX to PSTN | Ensure that calls are delivered from PBX to PSTN when the first trunk has failed on the LAN side | 1. 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 | |
| g711-017 | Failover of 1st trunk LAN - PSTN to PBX | Ensure that calls are delivered from PSTN to PBX when the first trunk has failed on the LAN side | 1. 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 | |
| g711-018 | Restore 1st trunk LAN - PBX to PSTN | Ensure that calls are delivered from PBX to PSTN when the first trunk has has been restored | 1. 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 | |
| g711-019 | Restore 1st trunk LAN - PSTN to PBX | Ensure that calls are delivered from PSTN to PBX when the first trunk has has been restored | 1. 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 | |
| g711-020 | Failover of 2nd trunk LAN - PBX to PSTN | Ensure that calls are delivered from PBX to PSTN when the second trunk has failed on the LAN side | 1. 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 | |
| g711-021 | Failover of 2nd trunk LAN - PSTN to PBX | Ensure that calls are delivered from PSTN to PBX when the second trunk has failed on the LAN side | 1. 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 | |
| g711-022 | Restore 2nd trunk LAN - PBX to PSTN | Ensure that calls are delivered from PBX to PSTN when the second trunk has has been restored | 1. 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 | |
| g711-023 | Restore 2nd trunk LAN - PSTN to PBX | Ensure that calls are delivered from PSTN to PBX when the second trunk has has been restored | 1. 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.
Overview
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