Cisco Systems 1.3.0 User Manual

VOCALVovida Open Communication Application LibrarySystem Administration GuideSoftware Version 1.3.0

1-4 Working With The GUI EnvironmentItems and Fields Table 1-2 describes the items found on the Login Screen.Password AdministrationThere is a separat

C-44 SIP IP Phone to SIP IP Phone: Forward All Callssip-req: ACK sip:[email protected]:5060 SIP/2.0 [192.168.26.180:5060->192.168.26.12:5

Phone to SIP Phone via Gateway: Call ScreeningC-45Phone to SIP Phone via Gateway: Call ScreeningCall Scenario Figure C-16 illustrates the following ca

C-46Phone to SIP Phone via Gateway: Call Screening Figure C-17. Call Flow Diagram: Call ScreeningSIP Phone UA Marshal1. INVITE2. 100Redirect Server3.

Phone to SIP Phone via Gateway: Call ScreeningC-47Call Trace The following call trace shows a call, originating from an on-network SIP IP phone, being

C-48 Phone to SIP Phone via Gateway: Call ScreeningHeader: a=fmtp:101 0-11------------------------------------------------------------

Phone to SIP Phone via Gateway: Call ScreeningC-49Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2Header: Via: SI

C-50 Phone to SIP Phone via Gateway: Call ScreeningHeader: Content-Length: 0----------------------------------------------------------

SIP Phone to PSTN: Call BlockingC-51SIP Phone to PSTN: Call BlockingCall Scenario Figure C-16 illustrates the following call scenario:• User A initiat

C-52SIP Phone to PSTN: Call BlockingFigure C-19. Call Flow DIagram: SIP IP Phone to PSTN: Call BlockingSIP Phone UA Marshal1. INVITE2. 100Redirect Ser

SIP Phone to PSTN: Call BlockingC-53Call Trace The following call trace shows a call, originating from an on-network SiP IP phone, being blocked by th

Working With The GUI Environment1-5Overview of the User Configuration ScreenIntroduction This section describes the buttons, option boxes, and data fi

C-54 SIP Phone to PSTN: Call BlockingHeader: a=rtpmap:101 telephone-event/8000Header: a=fmtp:101 0-11-----------------

SIP Phone to PSTN: Call BlockingC-55sip-res: SIP/2.0 403 Forbidden [192.168.26.220:6072->192.168.26.180:6060]Header: Via: SIP/2.

C-56 SIP IP Phone to SIP IP Phone: Call ReturnSIP IP Phone to SIP IP Phone: Call ReturnCall Scenario Figure C-20 illustrates the following call scenar

C-57SIP IP Phone to SIP IP Phone: Call ReturnFigure C-21. SIP IP Phone to SIP IP Phone: Call Return — Diagram 1SIP Phone UA Marshal1. INVITE2. 100Redi

C-58SIP IP Phone to SIP IP Phone: Call ReturnFigure C-22. SIP IP Phone to SIP IP Phone: Call Return — Diagram 2SIP Phone UA Marshal Redirect ServerFea

C-59SIP IP Phone to SIP IP Phone: Call ReturnFigure C-23. SIP IP Phone to SIP IP Phone: Call Return — Diagram 3SIP Phone UA Marshal Redirect ServerFea

C-60SIP IP Phone to SIP IP Phone: Call ReturnFigure C-24. SIP IP Phone to SIP IP Phone: Call Return — Diagram 4SIP Phone UA Marshal Redirect ServerFea

SIP IP Phone to SIP IP Phone: Call ReturnC-61Call Trace The following call trace shows a call return request leading to an established call between tw

C-62 SIP IP Phone to SIP IP Phone: Call ReturnHeader: a=fmtp:101 0-11-----------------------------------------------------------------

SIP IP Phone to SIP IP Phone: Call ReturnC-63Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2Header: Via: SIP/2.0

1-6 Working With The GUI EnvironmentAfter Data Entry Figure 1-3 shows what the screen looks like after some users have been added. For more informatio

C-64 SIP IP Phone to SIP IP Phone: Call Return----------------------------------------------------------------- SDP Headers----

SIP IP Phone to SIP IP Phone: Call ReturnC-65Header: Contact: <sip:[email protected]:5060>Header: Content-Lengt

C-66 SIP IP Phone to SIP IP Phone: Call ReturnHeader: To: <sip:[email protected]:5060>;tag=c29430001e2620-0Header:

SIP IP Phone to SIP IP Phone: Call ReturnC-67Header: To: <sip:[email protected]:5060>Header: Call-ID: c3943000

C-68 SIP IP Phone to SIP IP Phone: Call ReturnHeader: User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabledHeader: Accept: a

SIP IP Phone to SIP IP Phone: Call ReturnC-69-----------------------------------------------------------------sip-req: ACK sip:*[email protected]:5

C-70 SIP IP Phone to SIP IP Phone: Call ReturnHeader: To: <sip:*[email protected]:5060>Header: Call-ID: c2943000

SIP IP Phone to SIP IP Phone: Call ReturnC-71sip-req: INVITE sip:[email protected]:5060;user=phone SIP/2.0 [192.168.26.180:5060->192.168.26.20

C-72 SIP IP Phone to SIP IP Phone: Call Return-----------------------------------------------------------------Header: v=0Header:

SIP IP Phone to SIP IP Phone: Call ReturnC-73Header: m=audio 29956 RTP/AVP 0 101Header: a=rtpmap:0 pcmu/8000Header:

Working With The GUI Environment1-7Option Boxes The option boxes filter the fields displayed on the User Configuration screen. If none of the boxes is

C-74 SIP IP Phone to SIP IP Phone: Call ReturnHeader: Route: <sip:[email protected]:5060;maddr=192.168.26.180>, <sip:6711@1

User Agent to User Agent: Call WaitingC-75User Agent to User Agent: Call WaitingCall Scenario Figure C-25 illustrates the following call scenario:• Us

C-76User Agent to User Agent: Call WaitingFigure C-26. User Agent to User Agent: Call Waiting — Diagram 1VOCAL User Agent UA Marshal1. INVITE2. 100Red

C-77User Agent to User Agent: Call WaitingFigure C-27. User Agent to User Agent: Call Waiting — Diagram 2VOCAL User Agent UA Marshal Redirect Server V

C-78User Agent to User Agent: Call WaitingFigure C-28. User Agent to User Agent: Call Waiting — Diagram 3VOCAL User Agent UA Marshal Redirect Server V

User Agent to User Agent: Call WaitingC-79Call Trace The following call trace shows a third party attempting to connect to a phone that is engaged in

C-80 User Agent to User Agent: Call WaitingHeader: F-----------------------------------------------------------------

User Agent to User Agent: Call WaitingC-81-----------------------------------------------------------------sip-req: ACK sip:[email protected]:506

C-82 User Agent to User Agent: Call WaitingHeader: From: UserAgent<sip:[email protected]:5060;user=phone>Header:

User Agent to User Agent: Call WaitingC-83Header: From: UserAgent<sip:[email protected]:5060;user=phone>Header:

1-8 Working With The GUI EnvironmentRight-Mouse-Click Menu OptionsTable 1-6 shows the options available from the right-mouse-click menu.Option Boxes I

C-84 User Agent to User Agent: Call WaitingHeader: m=audio 60335 RTP/AVP 0Header: a=rtpmap:0 PCMU/8000Header:

User Agent to User Agent: Call WaitingC-85Header: Content-Length: 0-----------------------------------------------------------------

C-86 User Agent to User Agent: Call WaitingHeader: Route: <sip:[email protected]:5060;maddr=192.168.66.180>,<sip:[email protected]

User Agent to User Agent: Call WaitingC-87sip-res: SIP/2.0 200 OK [192.168.66.180:5060->192.168.66.3:5060]Header:

C-88 User Agent to User Agent: Call Waiting SDP Headers-----------------------------------------------------------------Header:

User Agent to User Agent: Call WaitingC-89Header: Content-Type: application/sdpHeader: Content-Type: application/sdpHe

C-90 User Agent to User Agent: Call Waitingsip-req: BYE sip:[email protected]:5060;maddr=192.168.66.180 SIP/2.0 [192.168.66.2:5060->192.168.66

SIP IP Phone to SIP IP Phone: Forward to Voice MailC-91SIP IP Phone to SIP IP Phone: Forward to Voice MailCall Scenario Figure C-29 illustrates the fo

C-92SIP IP Phone to SIP IP Phone: Forward to Voice MailFigure C-30. SIP IP Phone to SIP IP Phone: Forward to Voice Mail — Diagram 1SIP Phone UA Marsha

C-93SIP IP Phone to SIP IP Phone: Forward to Voice MailFigure C-31. SIP IP Phone to SIP IP Phone: Forward to Voice Mail — Diagram 2SIP Phone UA Marsha

Adding, Viewing, Editing, and Deleting Users1-9Adding, Viewing, Editing, and Deleting UsersIntroduction The ”Working With The GUI Environment” section

C-94SIP IP Phone to SIP IP Phone: Forward to Voice MailFigure C-32. SIP IP Phone to SIP IP Phone: Forward to Voice Mail — Diagram 3SIP Phone UA Marsha

SIP IP Phone to SIP IP Phone: Forward to Voice MailC-95Call Trace The following call trace shows a SIP IP phone attempting to call another on-network

C-96 SIP IP Phone to SIP IP Phone: Forward to Voice MailHeader: t=0 0Header: m=audio 23994 RTP/AVP 0 101Header:

SIP IP Phone to SIP IP Phone: Forward to Voice MailC-97 SIP Headers------------------------------------------------------------

C-98 SIP IP Phone to SIP IP Phone: Forward to Voice Mail-----------------------------------------------------------------Header: v=0He

SIP IP Phone to SIP IP Phone: Forward to Voice MailC-99Header: a=rtpmap:0 pcmu/8000Header: a=rtpmap:101 telephone-even

C-100 SIP IP Phone to SIP IP Phone: Forward to Voice MailHeader: Via: SIP/2.0/UDP 192.168.56.220:5074;branch=102Header:

SIP IP Phone to SIP IP Phone: Forward to Voice MailC-101Header: Contact: <sip:[email protected]:5082>Header: R

C-102 SIP IP Phone to SIP IP Phone: Forward to Voice MailHeader: Route: <sip:192.168.56.220:5060;maddr=192.168.56.220>,<sip:6

User Agent to User Agent: Consulted TransferC-103User Agent to User Agent: Consulted TransferCall Scenario Figure C-33 illustrates the following call

1-10 Adding, Viewing, Editing, and Deleting UsersAdding New Users Introduction This section describes how to add new users.Procedure: Adding a New Use

C-104User Agent to User Agent: Consulted TransferFigure C-34. User Agent to User Agent: Consulted Transfer — Diagram 1VOCAL User Agent UA Marshal1. IN

C-105User Agent to User Agent: Consulted TransferFigure C-35. User Agent to User Agent: Consulted Transfer — Diagram 2VOCAL User Agent UA Marshal VOCA

C-106User Agent to User Agent: Consulted TransferFigure C-36. User Agent to User Agent: Consulted Transfer — Diagram 3VOCAL User Agent UA Marshal VOCA

C-107User Agent to User Agent: Consulted TransferFigure C-37. User Agent to User Agent: Consulted Transfer— Diagram 4VOCAL User Agent UA Marshal VOCAL

C-108 User Agent to User Agent: Consulted TransferCall Trace The following call trace shows a consulted call transfer between two SIP IP phones.------

User Agent to User Agent: Consulted TransferC-109Header: -----------------------------------------------------------------

C-110 User Agent to User Agent: Consulted Transfer SIP Headers-----------------------------------------------------------------

User Agent to User Agent: Consulted TransferC-111Header: c=IN IP4 0.0.0.0Header: t=3174939344 0Header:

C-112 User Agent to User Agent: Consulted Transfer-----------------------------------------------------------------sip-req: ACK sip:[email protected]

User Agent to User Agent: Consulted TransferC-113Header: o=- 1113249245 1113249245 IN IP4 192.168.66.1Header: s=VOVIDA

Adding, Viewing, Editing, and Deleting Users1-11Adding Users: Administrator’s Edit User ScreenEdit User Screen Figure 1-5 illustrates the edit user sc

C-114 User Agent to User Agent: Consulted TransferHeader: Record-Route: <sip:[email protected]:5060;maddr=192.168.26.180>,<

User Agent to User Agent: Consulted TransferC-115 SIP Headers-----------------------------------------------------------------s

C-116 User Agent to User Agent: Consulted TransferHeader: Contact: <sip:[email protected]:5060;user=phone>Header:

User Agent to User Agent: Consulted TransferC-117sip-req: TRANSFER sip:[email protected]:5060 SIP/2.0 [192.168.26.180:5060->192.168.66.2:5060]

C-118 User Agent to User Agent: Consulted TransferHeader: CSeq: 1 INVITEHeader: Subject: VovidaINVITEHeader:

User Agent to User Agent: Consulted TransferC-119Header: s=VOVIDA SessionHeader: c=IN IP4 192.168.66.3Header:

C-120 User Agent to User Agent: Consulted Transfer-----------------------------------------------------------------sip-req: BYE sip:[email protected]

User Agent to User Agent: Consulted TransferC-121Header: Content-Length: 0------------------------------------------------------------

C-122 User Agent to User Agent: Blind TransferUser Agent to User Agent: Blind TransferCall Scenario Figure C-38 illustrates a call scenario in which:•

C-123User Agent to User Agent: Blind TransferFigure C-39. User Agent to User Agent: Blind Transfer — Diagram 1VOCAL User Agent UA Marshal1. INVITE2. 1

1-12 Adding, Viewing, Editing, and Deleting UsersGroup This field is a text identifier to help you classify your users.Marshal GroupAllows you to sele

C-124User Agent to User Agent: Blind TransferFigure C-40. User Agent to User Agent: Blind Transfer — Diagram 2VOCAL User Agent UA Marshal VOCAL User A

C-125User Agent to User Agent: Blind TransferFigure C-41. User Agent to User Agent: Blind Transfer — Diagram 3VOCAL User Agent UA Marshal VOCAL User A

C-126User Agent to User Agent: Blind TransferFigure C-42. User Agent to User Agent: Blind Transfer — Diagram 4VOCAL User Agent UA Marshal VOCAL User A

User Agent to User Agent: Blind TransferC-127Call Trace The following call trace shows an unconsulted call transfer.----------------------------------

C-128 User Agent to User Agent: Blind Transfer----------------------------------------------------------------- SIP Headers----

User Agent to User Agent: Blind TransferC-129Header: s=VOVIDA SessionHeader: c=IN IP4 192.168.66.2Header:

C-130 User Agent to User Agent: Blind Transfer----------------------------------------------------------------- SIP Headers----

User Agent to User Agent: Blind TransferC-131Header: c=IN IP4 192.168.66.2Header: t=3174939460 0Header:

C-132 User Agent to User Agent: Blind Transfer-----------------------------------------------------------------sip-req: INVITE sip:[email protected].

User Agent to User Agent: Blind TransferC-133Header: s=VOVIDA SessionHeader: c=IN IP4 0.0.0.0Header: t

Adding, Viewing, Editing, and Deleting Users1-13JTAPI Check the Enabled option box to enable the JTAPI feature. With this feature enabled the user can

C-134 User Agent to User Agent: Blind Transfer----------------------------------------------------------------- SIP Headers----

User Agent to User Agent: Blind TransferC-135Header: o=- 1986829226 1986829226 IN IP4 192.168.66.2Header: s=VOVIDA Ses

C-136 User Agent to User Agent: Blind Transfer----------------------------------------------------------------- SIP Headers----

User Agent to User Agent: Blind TransferC-137 SIP Headers-----------------------------------------------------------------sip-r

C-138 User Agent to User Agent: Blind TransferHeader: ÷Ï-----------------------------------------------------------------

User Agent to User Agent: Blind TransferC-139sip-req: BYE sip:[email protected]:5060 SIP/2.0 [192.168.26.180:5060->192.168.66.3:5060]Head

C-140 JTAPIJTAPICall Scenario Figure C-43 illustrates the following call scenario:• A user uses a JTAPI User Agent on a PC to remotely instruct SIP Ph

C-141JTAPIFigure C-44. Call Flow Diagram: JTAPI — Diagram 1JTAPI User Agent Redirect Server1. INVITE2. 3023. ACKUA Marshal4. INVITE5. 1006. INVITE7. 3

C-142JTAPIFigure C-45. Call Flow Diagram: JTAPI — Flow Diagram 2JTAPI User Agent Redirect Server UA Marshal VOCAL User Agent SIP Phone16. 30217. 20018

C-143JTAPIFigure C-46. Call Flow Diagram: JTAPI — Flow Diagram 3JTAPI User Agent Redirect Server UA Marshal VOCAL User Agent SIP Phone32. 10033. 18034

ii Copyright Copyright © 2001, Cisco Systems, Inc.Guide Versions The following table matches the software versions with the guide versions:Version Thi

1-14 Adding, Viewing, Editing, and Deleting UsersCall Screening Option BoxCheck the Enabled option box to enable the Call Screening feature for the us

C-144JTAPIFigure C-47. Call Flow Diagram: JTAPI — Flow Diagram 4JTAPI User Agent Redirect Server UA Marshal VOCAL User Agent SIP Phone48. ACK49. ACK50

JTAPIC-145Call Trace The following call trace shows third party call control through a JTAPI server.--------------------------------------------------

C-146 JTAPIHeader: Content-Length: 0----------------------------------------------------------------- SIP Heade

JTAPIC-147Header: Via: SIP/2.0/UDP 192.168.5.11:25060;branch=301Header: From: <sip:[email protected]:25060>Header:

C-148 JTAPI SDP Headers-----------------------------------------------------------------Header: v=0Header:

JTAPIC-149Header: m=audio 23456 RTP/AVP 0Header: a=rtpmap:0 PCMU/8000Header: a=ptime:20---------------

C-150 JTAPIHeader: Call-ID: [email protected]: CSeq: 2 TRANSFERHeader: Content-Lengt

JTAPIC-151Header: Via: SIP/2.0/UDP 192.168.5.11:15060;branch=202Header: From: UserAgent<sip:[email protected]:5060

C-152 JTAPIsip-req: ACK sip:[email protected]:5060;user=phone SIP/2.0 [192.168.26.180:5060->192.168.26.200:5060]Header: Via: S

JTAPIC-153sip-res: SIP/2.0 200 OK [192.168.22.36:5060->192.168.26.180:5060]Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4

Adding, Viewing, Editing, and Deleting Users1-15Call Return Option BoxCheck the Call Return option box to enable the Call Return feature for the user.

C-154 JTAPIHeader: From: UserAgent<sip:[email protected]:5060>Header: To: 6711<sip:[email protected]:5060;

JTAPIC-155Header: o=CiscoSystemsSIP-IPPhone-UserAgent 26487 28247 IN IP4 192.168.26.10Header: s=SIP CallHeader:

C-156 JTAPIHeader: CSeq: 1 ACKHeader: Route: <sip:[email protected]:5060>Header: Proxy-Authoriz

JTAPIC-157Header: CSeq: 2 BYEHeader: Route: <sip:[email protected]:15060>Header: Content-Length: 0

C-158 JTAPIHeader: From: 6711<sip:[email protected]:5060;user=phone>;tag=c29430003e2620-0Header: To: UserAgent

Ad Hoc Conference Call Between User AgentsC-159Ad Hoc Conference Call Between User AgentsCall Scenario Figure C-48 illustrates the following call scen

C-160Ad Hoc Conference Call Between User AgentsFigure C-49. User Agent to User Agent to User Agent: Ad Hoc Conference Call: Disconnect Last Party — Di

C-161Ad Hoc Conference Call Between User AgentsFigure C-50. User Agent to User Agent to User Agent: Ad Hoc Conference Call: Disconnect Last Party — Di

C-162Ad Hoc Conference Call Between User AgentsFigure C-51. User Agent to User Agent to User Agent: Ad Hoc Conference Call: Disconnect Last Party — Di

C-163Ad Hoc Conference Call Between User AgentsFigure C-52. User Agent to User Agent to User Agent: Ad Hoc Conference Call: Disconnect Last Party — Di

1-16 Adding, Viewing, Editing, and Deleting UsersViewing Users: IndividuallyIntroduction This section describes how to view records for individual use

C-164Ad Hoc Conference Call Between User AgentsFigure C-53. User Agent to User Agent to User Agent: Ad Hoc Conference Call: Disconnect Last Party — Di

Ad Hoc Conference Call Between User AgentsC-165Call Trace The following call trace shows an ad hoc conference call between three users.---------------

C-166 Ad Hoc Conference Call Between User Agentssip-res: SIP/2.0 302 Moved Temporarily [192.168.46.200:5060->192.168.46.180:5060

Ad Hoc Conference Call Between User AgentsC-167Header: From: seymour<sip:[email protected]:5060;user=phone>Header:

C-168 Ad Hoc Conference Call Between User AgentsHeader: Record-Route: <sip:[email protected]:5060;maddr=192.168.46.180>,<si

Ad Hoc Conference Call Between User AgentsC-169================================================================= First call establi

C-170 Ad Hoc Conference Call Between User AgentsHeader: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=4Header: Via: SIP/

Ad Hoc Conference Call Between User AgentsC-171----------------------------------------------------------------- SIP Headers---

C-172 Ad Hoc Conference Call Between User AgentsHeader: To: 5202<sip:[email protected]:5060;user=phone>Header:

Ad Hoc Conference Call Between User AgentsC-173----------------------------------------------------------------- SIP Headers---

Adding, Viewing, Editing, and Deleting Users1-17Screen Capture: Viewing A Single UserFigure 1-6 illustrates selecting the data for a single user. Figu

C-174 Ad Hoc Conference Call Between User Agents SDP Headers-----------------------------------------------------------------He

Ad Hoc Conference Call Between User AgentsC-175sip-req: INVITE sip:[email protected]:5060;maddr=192.168.46.180 SIP/2.0 [192.168.46.1:5060->192

C-176 Ad Hoc Conference Call Between User AgentsHeader: Content-Length: 168-----------------------------------------------------------

Ad Hoc Conference Call Between User AgentsC-177Header: Via: SIP/2.0/UDP 192.168.46.1:5060Header: From: seymour<sip:

C-178 Ad Hoc Conference Call Between User AgentsHeader: t=3177798665 0Header: m=audio 23456 RTP/AVP 0Header:

Ad Hoc Conference Call Between User AgentsC-179----------------------------------------------------------------- SIP Headers---

C-180 Ad Hoc Conference Call Between User Agents-----------------------------------------------------------------Header: v=0Header:

Ad Hoc Conference Call Between User AgentsC-181sip-req: BYE sip:[email protected]:5060 SIP/2.0 [192.168.46.180:5060->192.168.46.3:5060]He

C-182 Ad Hoc Conference Call Between User AgentsHeader: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=4Header: Via: SIP/

Ad Hoc Conference Call Between User AgentsC-183----------------------------------------------------------------- SIP Headers---

1-18 Adding, Viewing, Editing, and Deleting UsersScreen Capture: Viewing Small Groups of UsersFigure 1-7 illustrates selecting the data for a small gr

C-184 Ad Hoc Conference Call Between User Agents

Symbols*69call flow C-56Numerics1xx and 2xx B-33xx Responses B-34xx Responses B-35xx Responses B-46xx Responses B-4900 # Admin Block 1-20900 # User Bl

Index-2Index (Continued)Long Distance Admin Block 1-20Long Distance User Block 1-22Marshal 1-19Name 1-19Password 1-21Static Reg Enabled 1-21Terminatin

Index-3Index (Continued)RREGISTER B-2Registration C-3Right mouse click menu 1-16SScreenEdit UserAdministrator view 1-11User data view 1-31Login 1-3SNM

Index-4Index (Continued)Usersadding new users 1-10deleting 1-27edit multiple users 1-28finding users 1-26Load all users 1-24–1-27procedure for editing

Adding, Viewing, Editing, and Deleting Users1-19Viewing Users: Data Fields DescriptionsIntroduction Different data fields appear in the user configura

1-20 Adding, Viewing, Editing, and Deleting UsersForward Busy/No Ans. GroupIndicates the name of the ForwardBusyNoAnswer Feature server group.Failure

Adding, Viewing, Editing, and Deleting Users1-21Call Return Enabled Indicates whether the Call Return feature is enabled for the user:• Deselected: in

1-22 Adding, Viewing, Editing, and Deleting UsersUser Data Field When the Show user data option box is checked, these data field appear in addition to

Adding, Viewing, Editing, and Deleting Users1-23900 # User Block Indicates whether 900 Number Block feature is set by the user:• Deselected: indicates

3UHIDFHIntroduction This chapter is a general introduction to the System Administration manual, and provides information about the intentions and orga

1-24 Adding, Viewing, Editing, and Deleting UsersViewing Users: All UsersIntroduction This section explains how to use the Load all users button and t

Adding, Viewing, Editing, and Deleting Users1-25Load All Users Figure 1-8 shows the use of the Load all users button.Figure 1-8. User Configuration Sc

1-26 Adding, Viewing, Editing, and Deleting UsersFinding UsersIntroduction You can highlight any of the users by clicking their record with the mouse.

Adding, Viewing, Editing, and Deleting Users1-27Deleting UsersDeleting User To delete a user or multiple users, follow these steps:Table 1-14. Procedu

1-28 Adding, Viewing, Editing, and Deleting UsersEditing Users: Administrator ControlledIntroduction This section describes how to edit users.Procedur

Adding, Viewing, Editing, and Deleting Users1-29Editing User: Show Alias Introduction The aliases names associated with each users can be displayed us

1-30 Adding, Viewing, Editing, and Deleting UsersEditing User Features: User ControlledIntroduction The VOCAL system provides a web page for users to

Adding, Viewing, Editing, and Deleting Users1-31Editing User Feature: Edit User ScreenShow User Data ViewFigure 1-10 illustrates the edit user screen

1-32 Adding, Viewing, Editing, and Deleting UsersAliases This field displays aliases associated with this user. To add aliases for the user:1) Right m

Adding, Viewing, Editing, and Deleting Users1-33Call Screening The user can screen a call by name and number. To add numbers to screen:1) Right mouse

iv Documentation ConventionsThe following is a list of conventions used in this guide:Additional resourcesPublicationsAn Installation Guide, which inc

1-34 Adding, Viewing, Editing, and Deleting Users

1HWZRUN0DQDJHPHQWThis chapter describes network management and statistics for the VOCAL system.Topic See PageSNMP Support . . . . . . . . . . . . .

2-2 SNMP SupportSNMP Support Overview VOCAL supports Simple Network Management Protocol (SNMP) monitoring from:• the VOCAL SNMP GUI - this supports mo

SNMP Support2-3MIBsIntroduction In a TCP/IP-based network, each device maintains a set of variables describing its state. In Simple Network Management

2-4 SNMP SupportVOCAL Enterprise MIBFor more information refer to the /usr/local/vocal/proxies/netMgnt directory:• VOVIDA-LOCAL-GRP-MIB.txt• VOVIDA-NO

SNMP Support2-5VOCAL SNMP GUIServer Status MonitoringEach VOCAL system server sends (via multicast) heartbeat packets to its peers at a predefined int

2-6 SNMP SupportVOCAL SNMP GUI Screen ElementsHosts & Processes This frame displays the host server and indicates whether they are active (blue) o

)HDWXUHVThis chapter describes features supported by the VOCAL system.Topic See PageFeatures. . . . . . . . . . . . . . . . . . . . . . . . . . . . .

A-2 FeaturesFeaturesIntroduction This section describes the types of feature supported by the VOCAL system.Overview The VOCAL system supports two type

Core System FeaturesA-3Core System FeaturesTypes of Core System FeaturesThere are two types of system features—calling features and called features. T

Table of ContentsvPrefaceChapter 1. Setting Up UsersWorking With The GUI Environment . . . . . . . . . . . . . . . . . . . . . . . . . 1-2Adding, Vie

A-4 Core System FeaturesCall Return Call return allows the user to call back the last caller. The user dials *69 to dial up the last caller.Call Scree

Set-Based FeaturesA-5Set-Based FeaturesDefinition Set based features are features that a user can enable from a phone set. These features are an examp

A-6 Set-Based FeaturesAdHoc ConferencingThe Conference key on a phone set allows the user to set up a conference call with a number of people. To set

6XSSRUWHG6,30HVVDJHVTopic See PageSIP Request Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-2SIP Response Mes

B-2 SIP Request MessagesSIP Request MessagesSupported SIP Request MessagesThe VOCAL system supports these SIP request messages:Table B-1. SIP Request

SIP Response MessagesB-3SIP Response MessagesSIP Response Messages CategoryThe VOCAL system supports all SIP response messages:• 1xx Responses - Infor

B-4 SIP Response Messages• 409 Conflict • 410 Gone • 411 Length Required • 413 Request Entity Too Large • 414 Request-URI Too Large • 415 Unsupported

&DOO)ORZVThis chapter provides call flows diagram and IP trace logs for several call scenarios.Topic See PageSIP Phone: Registration. . . . . . .

C-2 Topic (continued) See PageUser Agent to User Agent: Consulted Transfer. . . . . . . . . . . . . . . . C-103User Agent to User Agent: Blind Transf

SIP Phone: RegistrationC-3SIP Phone: RegistrationCall Scenario Figure C-1 illustrates a SIP phone registering with the Marshal server. Authentication

vi Table of Contents (continued)

C-4 SIP Phone: RegistrationRegistration: Access List AuthenticationCall Flow Diagram Figure C-2 shows a SIP IP phone registering with the Redirect ser

SIP Phone: RegistrationC-5Header: Expires: 3600Header: Content-Length: 0----------------------------------------------

C-6 SIP Phone: RegistrationRegistration: Digest AuthenticationCall Flow Diagram Figure C-3 shows a SIP IP phone registering with the Redirect server.

SIP Phone: RegistrationC-7Header: Authorization: Digest username=”6711”,realm=”vovida.com”,uri=”sip:192.168.26.180”,response=”fee2efef

C-8 SIP IP Phone to SIP IP Phone: Call Setup and DisconnectSIP IP Phone to SIP IP Phone: Call Setup and DisconnectCall Scenario Figure C-4 illustrates

C-9SIP IP Phone to SIP IP Phone: Call Setup and DisconnectFigure C-5. Call Flow Diagram: SIP Phone to SIP Phone — Diagram 1SIP Phone UA Marshal1. INVI

C-10SIP IP Phone to SIP IP Phone: Call Setup and DisconnectFigure C-6. Call Flow Diagram: SIP Phone to SIP Phone — Diagram 216. ACK17. BYE18. BYE19. 2

SIP IP Phone to SIP IP Phone: Call Setup and DisconnectC-11Call Trace The following call trace shows a successful call setup between two, on-network I

C-12 SIP IP Phone to SIP IP Phone: Call Setup and DisconnectHeader: a=fmtp:101 0-11---------------------------------------------------

SIP IP Phone to SIP IP Phone: Call Setup and DisconnectC-13-----------------------------------------------------------------sip-req: ACK sip:5120@1

6HWWLQJ8S8VHUVThis chapter describes how to add users to the system and how to maintain the user data base.Topic See PageWorking With The GUI Enviro

C-14 SIP IP Phone to SIP IP Phone: Call Setup and DisconnectHeader: To: <sip:[email protected]:5060>;tag=c29430002e0620-0Heade

SIP IP Phone to SIP IP Phone: Call Setup and DisconnectC-15Header: Call-ID: [email protected]:

C-16 SIP IP Phone to Analog Phone via GatewaySIP IP Phone to Analog Phone via GatewayCall Scenario Figure C-7 illustrates a SIP phone to analog phone

C-17SIP IP Phone to Analog Phone via GatewayFigure C-8. Call Flow Diagram: SIP IP Phone to SIP IP Phone via SIP Gateway — Diagram 1SIP Phone UA Marsha

C-18SIP IP Phone to Analog Phone via GatewayFigure C-9. Call Flow Diagram: SIP IP Phone to SIP IP Phone via SIP Gateway — Diagram 216. ACK17. ACK18. A

SIP IP Phone to Analog Phone via GatewayC-19Call Trace The following trace shows a call originating from an on-network SIP phone and being routed thro

C-20 SIP IP Phone to Analog Phone via GatewayHeader: a=rtpmap:101 telephone-event/8000Header: a=fmtp:101 0-11---------

SIP IP Phone to Analog Phone via GatewayC-21sip-req: INVITE sip:[email protected]:5060;user=phone SIP/2.0 [192.168.36.110:5060->192.168.16

C-22 SIP IP Phone to Analog Phone via GatewayHeader: Via: SIP/2.0/UDP 192.168.6.20:5060Header: From: <sip:5120@192.

SIP IP Phone to Analog Phone via GatewayC-23Header: Via: SIP/2.0/UDP 192.168.36.180:5060;branch=2Header: Via: SIP/2.0/

1-2 Working With The GUI EnvironmentWorking With The GUI EnvironmentOverview This section describes:• the login screen and how to log into the VOCAL s

C-24 SIP IP Phone to Analog Phone via GatewayHeader: CSeq: 100 ACKHeader: Route: <sip:[email protected]:5060&

SIP IP Phone to Analog Phone via GatewayC-25Header: Via: SIP/2.0/UDP 192.168.36.180:5060;branch=4,SIP/2.0/UDP 192.168.36.110:5060;bran

C-26 SIP Phone to Phone via Gateway: Called Party is BusySIP Phone to Phone via Gateway: Called Party is BusyCall Scenario Figure C-10 illustrates Use

C-27SIP Phone to Phone via Gateway: Called Party is BusyFigure C-11. Call Flow Diagram: SIP Phone to Phone: Called Party is Busy — Diagram 1SIP Phone

C-28SIP Phone to Phone via Gateway: Called Party is BusyFigure C-12. Call Flow Diagram: SIP Phone to Phone: Called Party is Busy — Diagram 2SIP Phone

SIP Phone to Phone via Gateway: Called Party is BusyC-29Call Trace The following call trace shows a call originating from an on-network SIP phone, bei

C-30 SIP Phone to Phone via Gateway: Called Party is BusyHeader: c=IN IP4 192.168.26.10Header: t=0 0Header:

SIP Phone to Phone via Gateway: Called Party is BusyC-31Header: Content-Length: 0-----------------------------------------------------

C-32 SIP Phone to Phone via Gateway: Called Party is Busy----------------------------------------------------------------- SIP

SIP Phone to Phone via Gateway: Called Party is BusyC-33----------------------------------------------------------------- SIP H

Working With The GUI Environment1-3Logging InIntroduction The Provisioning Login screen provides access for Administrators to work with the users, and

C-34 SIP Phone to Phone via Gateway: Called Party is BusyHeader: Via: SIP/2.0/UDP 192.168.26.10:5060Header: From: <

SIP IP Phone to SIP IP Phone: Forward All CallsC-35SIP IP Phone to SIP IP Phone: Forward All CallsCall Scenario Figure C-13 illustrates the following

C-36SIP IP Phone to SIP IP Phone: Forward All CallsFigure C-14. Call Flow Diagram: SIP IP Phone to SIP IP Phone: Forward All Calls — Diagram 1SIP Phon

C-37SIP IP Phone to SIP IP Phone: Forward All CallsFigure C-15. Call Flow Diagram: SIP IP Phone to SIP IP Phone: Forward All Calls — Diagram 2SIP Phon

C-38 SIP IP Phone to SIP IP Phone: Forward All CallsCall Trace The following call trace shows a call originating from an on-network SIP IP phone being

SIP IP Phone to SIP IP Phone: Forward All CallsC-39Header: a=rtpmap:101 telephone-event/8000Header: a=fmtp:101 0-11---

C-40 SIP IP Phone to SIP IP Phone: Forward All CallsHeader: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2Header: Via:

SIP IP Phone to SIP IP Phone: Forward All CallsC-41Header: m=audio 30224 RTP/AVP 0 101Header: a=rtpmap:0 pcmu/8000Head

C-42 SIP IP Phone to SIP IP Phone: Forward All Callssip-req: INVITE sip:[email protected]:5060 SIP/2.0 [192.168.26.180:5060->192.168.26.12:5

SIP IP Phone to SIP IP Phone: Forward All CallsC-43Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4,SIP/2.0/UDP 192.168.26.180:50