Instruction/ maintenance manual of the product 1.3.0 Cisco Systems
Go to page of 244
VOCAL V ovida Op en Communicatio n Application Libr ary System Administration G uide Softwar e V ersion 1. 3.0.
ii Copyright Copyri ght © 2001 , Cisco Sys tems, Inc. Guide V ersions The followi ng t able matche s the software versi ons wit h the guide versions : Ve r s i o n This manual is w ritt en to suppor t VOCAL V ersion 1.3. 0. Support The pr imary locat ion for support, informati on and a ssist ance for t he VOCAL system is http ://www .
3U HIDFH Introduct ion This chapt er i s a general introduc tion to the Syst em Administrati on manual , and provides inf or mation about the inten tions and organizat i on of the manual. It also provi des informati on about additiona l res ources avail abl e from http://www .
iv Document ation Conventions The followi ng is a l ist of convention s used in this guide: Additional resources Publica tions An Inst allat ion Guide , whic h includes a system overvi ew , inst alla tion instruc tions and informat i on about provis ioning the server s is also availab le from http://www .
T able of Conten t s v Preface Chapter 1. Setting Up Users Working W it h The GUI Enviro nment . . . . . . . . . . . . . . . . . . . . . . . . . 1-2 Adding, Viewing, Edi ting, and Deleti ng Users . . . . . . . . . . . . . . . . . . 1-9 Chapter 2. Network Management SNMP Support .
vi T able of Conten t s (continued).
6HWWLQJ8S8VHUV This chapter desc ri bes how to add users to the system and how to maintai n the user dat a base. T opic See Page Working With The GUI Environment. . . . . . . . . . . . . . . . . . . . . . . . . . 1-2 Logging I n . . . . . . . . .
1-2 Wo rking W ith The GUI Environment W orking With The GUI Environment Overview This sect i on describes: • the login screen a nd how to log into t he VOCAL sys tem • the user configurat ion screen and avai lable t he buttons, options boxes, and data fi elds.
Wor king Wit h The GUI Environment 1-3 Loggi ng In Introduct ion The Provis ion ing Login scre en provi des access for Adminis tr ators to work with th e users, and for T ec hnicians to work with t he servers . Definition The l ogin screen is a java-enabl ed graphical user interface (GUI) tha t runs in a web browser .
1-4 Wo rking W ith The GUI Environment Items and Fields T able 1-2 describes the items found on the Login Screen. Password Administrat ion There is a separ at e user inter fac e for changing pa s swords and adding or removing account s for admini strators and te chnicia n s.
Wor king Wit h The GUI Environment 1-5 Overview of the Us er Configuratio n Screen Introduct ion This sect ion desc ribes the butt ons, opt ion boxe s, and dat a fi elds on th e User Configurat ion Screen. Screen Capture Figures 1-2 and 1-3 show the User Confi gurat ion screen as it appears whe n you login.
1-6 Wo rking W ith The GUI Environment After Dat a Entry Fi gure 1-3 shows what the screen looks li ke af ter some users have been added. For more infor mation about adding user s, see “ Adding New Users ” on page 1-10 .
Wor king Wit h The GUI Environment 1-7 Option Boxes The optio n box es filt er t he fi eld s displ aye d on t he Use r Co nfigur ation scr een. If none of the boxes is sel ected, onl y the Name, User Group, IP and Marsha l fields appea r . If all of t he boxes are selected, th en all of the fields appear on the User Configurati on screen.
1-8 Wo rking W ith The GUI Environment Right-Mouse-Cl ick Menu Options T able 1-6 shows the options avai lable from the right -mouse-cli ck menu. Option Boxes If you select eit her Edit or New , you must select at least one opti on box as well.
Adding, V iewing, Edit ing, and Del eting Use rs 1-9 Adding, V iewing, Editing, and Deleti ng Users Introduct ion The ” W ork ing With Th e GUI Envi ronment ” sec tion dis cussed the GUI buttons, option boxes an d a right -mouse cli ck menu t hat enables adding, viewing, editi ng and deleting user s.
1-10 Adding, V iewing, Editing, and Dele ting Users Adding New Users Introduct ion This secti on describes how to add new users. Procedure: Addi ng a New User T o add a new user , f ollow thes e step s: T able 1-7. Adding New Users St ep Action 1 Select the Show ad min dat a option box.
Adding, V iewing, Edit ing, and Del eting Use rs 1-1 1 Adding Us ers: Administ rator ’ s Ed it Us er Scre en Edit User Screen Figure 1-5 illustr ates the edit user screen that appear s whe n the show administrat or dat a option box is checked. Figure 1- 5.
1-12 Adding, V iewing, Editing, and Dele ting Users Group This field is a text identifier t o help you classify your users. Marshal Group Allows yo u to s elect a User Agent Marshal server gr oup fr om the pull down menu.
Adding, V iewing, Edit ing, and Del eting Use rs 1-13 JT API Check t he Enable d option bo x to e nable the J T A PI feature. With t his featur e enabled the user can pla ce calls using a JT API User Agent.
1-14 Adding, V iewing, Editing, and Dele ting Users Call Screening Option Box Check the Enabled opt ion box t o enabl e the Call Scr eening feat ure for the user . Note For version 1. 3.0 of VOCAL, phone numbers entered for call screening must incl ude the ar ea code, regardless if t hey are local or long-dis ta nce phone numbers.
Adding, V iewing, Edit ing, and Del eting Use rs 1-15 Call Return Option Box Check the Call Retur n option bo x to enable the Cal l Retur n feature for the user . Pull Down Menu The pull do wn menu allows you t o select a Feature server gr oup from the pull down menu.
1-16 Adding, V iewing, Editing, and Dele ting Users V iewing Users: Ind ividually Introduct ion This sect ion des cribes how to view r ecords for indi vidual us ers.
Adding, V iewing, Edit ing, and Del eting Use rs 1-17 Screen Capture : Viewi ng A Single User Figure 1-6 illustr ates sele cti ng the dat a for a single user . Figure 1-6. Displ ayi ng Data for a Single User 1. Sele c t a re c o rd 2. Right mouse click, selec t View The data is displayed for the 3.
1-18 Adding, V iewing, Editing, and Dele ting Users Screen Capture : V iewing Small Group s of Users Figure 1-7 illustr ates sele cting the dat a for a small group of users .
Adding, V iewing, Edit ing, and Del eting Use rs 1-19 V iewin g Users: Dat a Fiel ds Descri ptions Introduct ion Dif ferent dat a fields appear in the user confi guration scr een depending on the option boxes sel ect ed.
1-20 Adding, V iewing, Editing, and Dele ting Users Forward Busy/No Ans. Group Indicates the name o f t he ForwardBus yNoAnswer Feature server gro up. Failure Case Indicates the number or address to for ward calls to there is a problem wit h cont acting the destination c alled party .
Adding, V iewing, Edit ing, and Del eting Use rs 1-21 Call Return Enabled Indicates whether the Call Retur n feature is enabled for t he user: • Deselected : in dicates t hat this fe ature is disable d for the use r . • Selected: indi cates tha t this fea ture is enabl ed for the user .
1-22 Adding, V iewing, Editing, and Dele ting Users User Dat a Field When the Show user dat a option box is checked, these data fi eld appear in addition to the default data fields: T able 1- 1 1.
Adding, V iewing, Edit ing, and Del eting Use rs 1-23 900 # User Block Indicates wheth er 900 Number Block feature i s set by the user : • Deselected: in dicates t hat 900 Number block is not en abled by the user . 90 0 numbers are not blocked and user can dial 900 numbers.
1-24 Adding, V iewing, Editing, and Dele ting Users V iewing Users: All Use rs Introduct ion This secti on explains how to use the Load all users but ton and the option boxes to view use r dat a. Overview For s ituations where you n eed to compare t he dat a between users, you can click the Load al l user s button.
Adding, V iewing, Edit ing, and Del eting Use rs 1-25 Load All Users Figure 1-8 shows the use of the Load all users butto n. Figure 1-8. User Confi gurat ion Screen: Loading All Users 1. Cl ick Load all 3. Use t he 2. Select options users The dat a is displa yed.
1-26 Adding, V iewing, Editing, and Dele ting Users Finding Users Introduct ion Y ou can highlight any of the users by cl icki ng thei r rec ord with the mouse.
Adding, V iewing, Edit ing, and Del eting Use rs 1-27 Deleting Users Deleting User T o delet e a user or multipl e users, foll ow these st eps: T able 1- 14. Procedure for Deleti ng Users St ep Action 1 T o select a user , lef t mouse click on a row in the t able.
1-28 Adding, V iewing, Editing, and Dele ting Users Editing Users: Admin istrator Controlled Introduct ion This secti on describes how to edit user s. Procedure: Edi ting a User T o edit a user , foll.
Adding, V iewing, Edit ing, and Del eting Use rs 1-29 Editing User: Sho w Alias Introduct ion The aliases names associ ated with each users can be disp layed using t he show alias op ti on box. User names with alias es appear in it alics. What ’ s an Alias? An alias is another addres s o r phone number by whic h a user can be reached.
1-30 Adding, V iewing, Editing, and Dele ting Users Editing User Features: User Contro lled Introduct ion The VOCAL system provides a web page for users to maint ai n some of their features.
Adding, V iewing, Edit ing, and Del eting Use rs 1-31 Editing User Feature: Edit Us er Screen Show User Dat a Vi ew Figure 1- 10 illustrate s the edit user screen tha t appears when Show user dat a option box checked and the edi t right mouse opti on is used.
1-32 Adding, V iewing, Editing, and Dele ting Users Aliases Thi s field disp lays aliases asso ciated wit h thi s user . T o add aliases for t he user: 1) Right mouse clic k over the Alias area and sele ct add. 2) T ype the ali as name for the user . T o remove aliases, right mouse clic k the alias name and selec t remove.
Adding, V iewing, Edit ing, and Del eting Use rs 1-33 Call Screening The user c an scree n a call by name and nu mber . T o add numbers to s creen: 1) Right mouse clic k near the name and numbe r box. Select Add. 2) A Screen Calls From di alog box appears .
1-34 Adding, V iewing, Editing, and Dele ting Users.
1HWZ RU N0DQD JHPHQW This chapter desc ri bes net work management and stat istics for the VOCAL system. T opic See Page SNMP Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-2 MIB s . . . . . . . . . .
2-2 SNMP Support SNMP Support Overview VOCAL support s Simple Networ k Ma nagement Pr otocol (SNMP) mon it ori ng from : • the VOCAL SNMP GUI - this support s monitori ng of VOCAL server stat us. • a third p arty SNMP network manager . SNMP V ersion VOCAL support s SNMP versi on 2 (R FCs 1441 to 1 452).
SNMP Support 2-3 MIBs Introduct ion In a TCP/IP-based network, each devic e maint ains a set of variabl es describi ng its st ate. I n Simple Network Management Prot ocol (SNMP), these variabl es are known as objec ts , but t hese object s d o not h old the s ame meaning as t hose withi n an ob ject-ori ented progr amming archit ecture.
2-4 SNMP Support VOCAL Enterprise MIB For mor e informat ion r efer to the /u sr/local /vocal/ proxies/n etMgnt di rectory: • VOVIDA-LOCAL-GRP-MIB.tx t • VOVIDA-NOTIFICA TION S-MIB.txt • VOVIDA-SERVERGRP-MIB. txt • VOVIDA-SOFTSWITCHST A TS-MIB.
SNMP Support 2-5 VOCAL SNMP GUI Server St atus Monitoring Each VOCAL syste m server send s (vi a multicas t) heart beat p acket s to it s peers at a predefi ned int erval. The Heartb eat Server monit or s the exchange of heart beat p acket s betwee n VOCAL servers and sends server st atus trap messages to the network management system.
2-6 SNMP Support VOCAL SNMP GUI Screen Elements Host s & Processes This fr ame displays the host server and indi cates whether they are active (blue) or i nactive (red) . If a host server cont ains several processes, it wil l display a red ball if one or more of the processe s is inactive.
)HD WXU HV This chapter desc ri bes features suppor ted by the VOCAL system. T opic See Page Features. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A-2 Core System Features . . . . . . . . . . . . . .
A-2 Features Features Introduct ion This sect ion descri bes t he types of feature supported by the VOCAL system. Overview The VOCAL sy stem suppor ts t wo types of feat ures — cor e syst em features and set-based feat ures.
Core System Features A-3 Core System Features T ypes of Core System Features There are two types of system features — calli ng features and call ed features. The call ing featur es are assigned t o the cal l origi nator . The ca lled feat ures are assi gned to the call ing desti nation.
A-4 Core System Feat ures Call Return Call r eturn all ows the user to call back the last call er . The user dials * 69 to dial up the last caller . Call Screening Call screening all ows the user to block cal ls from a list of numbers. For example, when an screened number call s the user , the caller will receive a busy signal .
Set-Based Features A-5 Set-Based Features Definition Set based f eatures ar e features that a user can enable from a phone set. These feat ures are an example of how SIP-ba sed networ ks are abl e to transfer much of i t s intel ligence t o it s end-po int s.
A-6 Set-Based Feat ures AdHoc Conferencing The Conference k ey on a phone set allows the user t o set up a confer ence call wi th a number of peop le. T o set up a confer ence cal l: • Call t he first p erson. Press t he conference button to place the f irst ca ller on hold.
6XSSRU WHG6,30HVVD JHV T opic See Page SIP Request Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-2 SIP Response Messages .
B-2 SIP Request Messages SIP Request Messages Supported SIP Request Messages The VOCAL system support s these SIP r equest messages: T able B-1. SIP Request Messages Descriptions SIP Request Messages Descriptions INVITE Indicat es that th e user or service is being invited t o p articip ate in a ses sion.
SIP Response Messages B-3 SIP Response Messages SIP Response Messages Category The VOCAL system support s all SI P respon se messages: • 1xx Responses - Infor mation Responses • 2xx Responses - Su.
B-4 SIP Response Messages • 409 Conflict • 410 Gone • 41 1 Length Requir ed • 413 Request Entity T oo Large • 414 Request- URI T oo Large • 415 Unsupported Media T ype • 420 Bad Extensio.
&DOO)OR ZV This chapter pr ovi des call flows di agram and IP tr a ce logs for several call scenarios. T opic See Page SIP Phone: Registrat ion. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C-3 Registrat ion: Acces s List Authentic ation .
C-2 T opic ( continued) See Page User Agent t o User Agent: Consul ted Tr ansfer. . . . . . . . . . . . . . . . C-103 User Agent to User Agent: Blind Trans fer . . . . . . . . . . . . . . . . . . . . C-122 JTAP I . . . . . . . . . . . . . . . . . . . .
SIP Phone: Registrati on C-3 SIP Phone: Registration Call Scenario Figur e C-1 illustr ates a SIP phone register ing with the Marshal ser ver . Authenticat ion Methods There are three regist ration methods, no aut henticati on, access list authentic ation or digest authentic ation.
C-4 SIP Phone: Regist ration Registration: Access List A uthentication Call Flow Diagram Fi gur e C-2 shows a SIP IP phone registering with the Redir ect ser ver . The User Agent Marshal ser ver is using the Access Li st authentica tion method. Figure C-2.
SIP Phone: Registrati on C-5 Header: Expires: 3600 Header: Content-Length : 0 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 200 OK [192.
C-6 SIP Phone: Regist ration Registration: Digest Authe ntication Call Flow Diagram Fi gur e C-3 shows a SIP IP phone registering with the Redir ect ser ver .
SIP Phone: Registrati on C-7 Header: Authorization: Digest username=”6711”,realm=”vovida.com”,uri =”sip:192.168.26.180”,response=” fee2efef60a99b4576c 0437947959deb”,nonce=”966645751”,algor ithm=MD5 Header: Contact: <sip: 6711@192.
C-8 SIP IP Phone to SI P IP Phone: Cal l Setup and Disconnect SIP IP Phone to SIP IP Phon e: Call Setup and Disconn ect Call Scenario Figur e C-4 illustr ates a call between two, on-net work SIP IP phones.
C-9 SIP IP Phone to SIP IP Phone: Call Setup and Disconnect Figure C-5. Call Flow Diagram: SIP Phone to SIP Phone — Diagram 1 SIP Phone UA Marshal 1. INVITE 2. 100 Redirect Server 3. INVITE 4. 302 5. ACK 6. INVITE 7. 302 8. ACK SIP Phone 9. INVITE 10.
C-10 SIP IP Phone to SIP IP Phone: Call Setup and Disconnect Figure C-6. Call Flow Diagram: SIP Phone to SIP Phone — Diagram 2 16. ACK 17. BYE 18. BYE 19.
SIP IP Phone to SIP IP Phone: Call Setup and Disconnect C-1 1 Call T r ace The followi ng call trace shows a success ful call setup between two, on- network I P phones.
C-12 SIP IP Phone to SI P IP Phone: Cal l Setup and Disconnect Header: a=fmtp:101 0-1 1 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 302 Moved Temporar ily [192.
SIP IP Phone to SIP IP Phone: Call Setup and Disconnect C-13 -------------------------------------- --------------------------- sip-req: ACK sip:5120@192.168.36.20 0:5060;user=phone SIP/2.0 [192.1 68.36.180:5060- >192.168.36.200:5060] Header: Via: SIP/2.
C-14 SIP IP Phone to SI P IP Phone: Cal l Setup and Disconnect Header: To: <sip:5120@ 192.168.36.180:5060>;tag=c294300 02e0620-0 Header: Call-ID: c2943 000-e0563-2a1ce-2e323931@192.168 .6.21 Header: CSeq: 100 INVI TE Header: Server: Cisco IP Phone/ Rev.
SIP IP Phone to SIP IP Phone: Call Setup and Disconnect C-15 Header: Call-ID: c2943 000-e0563-2a1ce-2e323931@192.168 .6.21 Header: Route: <sip:51 20@192.
C-16 SIP IP Phone to Analog Phone via Gat eway SIP IP Phone to Analo g Phone via Gateway Call Scenario Figur e C-7 illustr ates a SIP phone to analog phone call made over an IP network vi a a ga teway .
C-17 SIP IP Phone to Analog Phone via Gateway Figure C-8. Call Flow Diagram: SIP IP Phone to SIP IP Phone via SIP Gateway — Diagr am 1 SIP Phone UA Marshal 1. INVITE 2. 100 Redirect Server 3. INVITE 4. 302 5. ACK 5300 Marshal 6. INVITE 7. 100 Cisco 5300 8.
C-18 SIP IP Phone to Analog Phone via Gateway Figure C-9. Call Flow Diagram: SIP IP Phone to SIP IP Phone via SIP Gateway — Diagr am 2 16. ACK 17. ACK 18.
SIP IP Pho ne to Analog Phone via Gateway C-19 Call T r ace The foll owing tra ce shows a cal l ori ginating from an on-network SI P phone and being routed thr ough a gateway to the PSTN.
C-20 SIP IP Phone to Analog Phone via Gat eway Header: a=rtpmap:101 t elephone-event/8000 Header: a=fmtp:101 0-1 1 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.
SIP IP Pho ne to Analog Phone via Gateway C-21 sip-req: INVITE sip:93831073@192.16 8.16.210:5060;user=phone SIP/2.0 [192.168.36.110:5060->192.168.16.210:5 060] Header: Via: SIP/2.0/U DP 192.168.36.110:5060;branch=4 Header: Via: SIP/2.0/U DP 192.168.
C-22 SIP IP Phone to Analog Phone via Gat eway Header: Via: SIP/2.0/U DP 192.168.6.20:5060 Header: From: <sip:512 0@192.168.6.20:5060> Header: To: <sip:93831 073@192.168.36.180:5060> Header: Call-ID: c2943 000-23e062-2e278-2e323931@192.16 8.
SIP IP Pho ne to Analog Phone via Gateway C-23 Header: Via: SIP/2.0/U DP 192.168.36.180:5060;branch=2 Header: Via: SIP/2.0/U DP 192.168.6.20:5060 Header: From: <sip:512 0@192.168.6.20:5060> Header: To: <sip:93831 073@192.168.36.180:5060>;tag=1AF 49448-1D50 Header: Call-ID: c2943 000-23e062-2e278-2e323931@192.
C-24 SIP IP Phone to Analog Phone via Gat eway Header: CSeq: 100 ACK Header: Route: <sip:93 831073@192.168.16.210:5060> Header: Proxy-Authoriz ation: Basic VovidaClassXSwitch Header: Content-Len.
SIP IP Pho ne to Analog Phone via Gateway C-25 Header: Via: SIP/2.0/U DP 192.168.36.180:5060;branch=4, SIP/2.0/UDP 192.168.36.110:5060;branch=2,SIP/2.0/U DP 192.168.16.210:50110 Header: From: <sip:938 31073@192.168.36.180:5060>;tag=1 AF49448-1D50 Header: To: <sip:5120@ 192.
C-26 SIP Phone to Phone via Gateway: Called Party is Busy SIP Phone to Phone vi a Gateway: Called Party is Busy Call Scenario Figur e C-10 illustr ates User A initi ati ng a call to User B while User B is busy .
C-27 SIP Phone to Phone via Gateway: Called Party is Busy Figure C-1 1. Call Flow Diagram: SIP Phone to Phone: Called Party is Busy — Diagram 1 SIP Phone UA Marshal 1. INVITE 2. 100 Redirect Server 3. INVITE 4. 302 5. ACK 5300 Marshal 6. INVITE 7. 100 Cisco 5300 8.
C-28 SIP Phone to Phone via Gateway: Called Party is Busy Figure C-12. Call Flow Diagram: SIP Phone to Phone: Called Party is Busy — Diagram 2 SIP Phone UA Marshal Redirect Server 5300 Marshal Cisco 5300 16.
SIP Phone to Phone via Gateway: Called Party is Busy C-29 Call T r ace The followi ng call trace shows a call ori ginating fr om an on-network SIP phone, bei ng routed t hrough a gateway to the PSTN, and ret urning a busy signal.
C-30 SIP Phone to Phone via Gateway: Called Party is Busy Header: c=IN IP4 192.1 68.26.10 Header: t=0 0 Header: m=audio 26268 RTP/AVP 0 101 Header: a=rtpmap:0 pcm u/8000 Header: a=rtpmap:101 t elephon.
SIP Phone to Phone via Gateway: Called Party is Busy C-31 Header: Content-Length : 0 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:93831069@192.
C-32 SIP Phone to Phone via Gateway: Called Party is Busy -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 183 Session Progre ss [192.168.
SIP Phone to Phone via Gateway: Called Party is Busy C-33 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 200 OK [192.168.26 .110:5060->192.
C-34 SIP Phone to Phone via Gateway: Called Party is Busy Header: Via: SIP/2.0/U DP 192.168.26.10:5060 Header: From: <sip:671 1@192.168.26.10:5060> Header: To: <sip:93831 069@192.168.26.180:5060>;tag=25A 5AD44-1FE9 Header: Call-ID: c2943 000-3e262-e4dc-2e323931@192.
SIP IP Phone to SIP IP Phone: Forward All Calls C-35 SIP IP Phone to SIP IP Phon e: Forward All Calls Call Scenario Figur e C- 13 illu strates t he foll owing cal l scenari o: • User A ini tia t es .
C-36 SIP IP Phone to SIP IP Phone: Forward All Calls Figure C-14. Call Flow Diagram: SIP IP Phone to SIP IP Phone: Forward All Calls — Diagr am 1 SIP Phone UA Marshal 1. INVITE 2. 100 Redirect Server 3. INVITE 4. 302 5. ACK Feature Server: FAC 6. INVITE 7.
C-37 SIP IP Phone to SIP IP Phone: Forward All Calls Figure C-15. Call Flow Diagram: SIP IP Phone to SIP IP Phone: Forward All Calls — Diagr am 2 SIP Phone UA Marshal Redirect Server Feature Server: FAC SIP Phone 16. ACK 17. INVITE 18. 100 19. 180 20.
C-38 SIP IP Phone to SIP IP Phone: For ward All Cal ls Call T r ace The followi ng call trace shows a call ori ginating fr om an on-network SIP IP phone being forward ed to a cal l forwar d destinat ion.
SIP IP Phone to SIP IP Phone: Forward All Calls C-39 Header: a=rtpmap:101 t elephone-event/8000 Header: a=fmtp:101 0-1 1 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.
C-40 SIP IP Phone to SIP IP Phone: For ward All Cal ls Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/U DP 192.168.26.10:5060 Header: From: <sip:671 1@192.168.26.10:5060> Header: To: <sip:6715@ 192.168.26.180:5060> Header: Call-ID: c2943 000-ce262-1b5c2-2e323931@192.
SIP IP Phone to SIP IP Phone: Forward All Calls C-41 Header: m=audio 30224 RTP/AVP 0 101 Header: a=rtpmap:0 pcm u/8000 Header: a=rtpmap:101 t elephone-event/8000 Header: a=fmtp:101 0-1 1 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.
C-42 SIP IP Phone to SIP IP Phone: For ward All Cal ls sip-req: INVITE sip:6716@192.168.26 .12:5060 SIP/2.0 [192.168.26. 180:5060- >192.168.26.12:5060] Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.
SIP IP Phone to SIP IP Phone: Forward All Calls C-43 Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=4, SIP/2.0/UDP 192.168.26.180:5060;branch=2,SIP/2.0/U DP 192.168.26.10:5060 Header: From: <sip:671 1@192.168.26.10:5060> Header: To: <sip:6716@ 192.
C-44 SIP IP Phone to SIP IP Phone: For ward All Cal ls sip-req: ACK sip:6716@192.168.26.12 :5060 SIP/2.0 [192.168.26. 180:5060- >192.168.26.12:5060] Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.
Phone to SIP Phone via Gateway: Call Screening C-45 Phone to SIP Phone via Gateway: Call Screening Call Scenario Figur e C- 16 illu strates t he foll owing cal l scenari o: • User A ini tia t es a c.
C-46 Phone to SIP Phone via Gateway: Call Screening Figure C-17. Call Flow Diagram: Call Screening SIP Phone UA Marshal 1. INVITE 2. 100 Redirect Server 3. INVITE 4. 302 5. ACK Feature Server: Call Screening 6. INVITE 7. 100 8. INVITE 9. 302 10. ACK 11.
Phone to SIP Phone via Gateway: Call Screening C-47 Call T r ace The foll owing call tr ace shows a c all, originat ing from an o n-network SIP IP phone, bei ng scree ned by the f eature ser ver .
C-48 Phone to SI P Phone via Gateway: Call Screening Header: a=fmtp:101 0-1 1 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 302 Moved Temporar ily [192.
Phone to SIP Phone via Gateway: Call Screening C-49 Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/U DP 192.168.26.11:5060 Header: From: <sip:671 5@192.168.26.11:5060> Header: To: <sip:6711@ 192.168.26.180:5060> Header: Call-ID: c3943 000-6978b-2995c-2e323931@192.
C-50 Phone to SI P Phone via Gateway: Call Screening Header: Content-Length : 0 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 403 Forbidden [192 .
SIP Phone t o PSTN: Call Blocking C-51 SIP Phone to PSTN: Call Blocking Call Scenario Figur e C- 16 illu strates t he foll owing cal l scenari o: • User A i nitiates a lo ng dist ance or 1-900 number call • The VOCAL System blocks the call Figure C-18.
C-52 SIP Phone to PSTN: Call Blocking Figure C-19. Call Flow DIagram: SIP IP Phone to PSTN: Call Blocking SIP Phone UA Marshal 1. INVITE 2. 100 Redirect Server 3.
SIP Phone t o PSTN: Call Blocking C-53 Call T r ace The foll owing call tr ace shows a c all, originat ing from an o n-network SiP IP phone, bei ng blocked by the feature ser ver .
C-54 SIP Phone to PSTN: Call Blocking Header: a=rtpmap:101 t elephone-event/8000 Header: a=fmtp:101 0-1 1 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.
SIP Phone t o PSTN: Call Blocking C-55 sip-res: SIP/2.0 403 Forbidden [192 .168.26.220:6072->192.168.26.180 :6060] Header: Via: SIP/2.0/U DP 192.168.26.180:6060;branch=2 Header: Via: SIP/2.0/U DP 192.168.26.12:5060 Header: From: <sip:671 5@192.168.
C-56 SIP IP Phone to SI P IP Phone: Cal l Retu rn SIP IP Phone to SIP IP Phon e: Call Return Call Scenario Figur e C- 20 illu strates t he foll owing cal l scenari o: • User A di als *69 t o deter mine the l ast numbe r that was called, User B • User A cal ls User B Figure C-20.
C-57 SIP IP Phone to SIP IP Phone: Call Return Figure C-21. SIP IP Phone to SIP IP Phone: Call Return — Diagr am 1 SIP Phone UA Marshal 1. INVITE 2. 100 Redirect Server 3. INVITE 4. 302 5. ACK Feature Server: Call Return 6. INVITE 7. 100 8. INVITE 9.
C-58 SIP IP Phone to SIP IP Phone: Call Return Figure C-22. SIP IP Phone to SIP IP Phone: Call Return — Diagr am 2 SIP Phone UA Marshal Redirect Server Feature Server: Call Return SIP Phone 16. INVITE 17. 100 18. 180 19. 180 20. 180 21. 180 22. CANCEL 23.
C-59 SIP IP Phone to SIP IP Phone: Call Return Figure C-23. SIP IP Phone to SIP IP Phone: Call Return — Diagr am 3 SIP Phone UA Marshal Redirect Server Feature Server: Call Return SIP Phone 32. INVITE 33. 302 34. ACK 35. INVITE 36. 100 37. 302 38. ACK 39.
C-60 SIP IP Phone to SIP IP Phone: Call Return Figure C-24. SIP IP Phone to SIP IP Phone: Call Return — Diagr am 4 SIP Phone UA Marshal Redirect Server Feature Server: Call Return SIP Phone 48. 180 49. 180 50. 200 51. 200 52. ACK 53. ACK 54. BYE 55.
SIP IP Phone to SIP I P Phone: Call Return C-61 Call T r ace The followi ng call tr ace shows a call ret urn req uest leadi ng to an esta blished call between two on- network SIP IP phones.
C-62 SIP IP Phone to SI P IP Phone: Cal l Retu rn Header: a=fmtp:101 0-1 1 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 302 Moved Temporar ily [192.
SIP IP Phone to SIP I P Phone: Call Return C-63 Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/U DP 192.168.26.11:5060 Header: From: <sip:671 5@192.168.26.11:5060> Header: To: <sip:6711@ 192.168.26.180:5060> Header: Call-ID: c3943 000-2978b-aa0e-2e323931@192.
C-64 SIP IP Phone to SI P IP Phone: Cal l Retu rn -------------------------------------- --------------------------- SDP Headers -------------------------------------- --------------------------- Header: v=0 Header: o=CiscoSystemsSIP-IPPhone-UserAgent 25077 6500 IN IP4 192.
SIP IP Phone to SIP I P Phone: Call Return C-65 Header: Contact: <sip: 6711@192.168.26.10:5060> Header: Content-Length : 0 Header: CC-Redirect: < sip:6711@192.
C-66 SIP IP Phone to SI P IP Phone: Cal l Retu rn Header: To: <sip:6711@ 192.168.26.180:5060>;tag=c294300 01e2620-0 Header: Call-ID: c3943 000-2978b-aa0e-2e323931@192.
SIP IP Phone to SIP I P Phone: Call Return C-67 Header: To: <sip:6711@ 192.168.26.180:5060> Header: Call-ID: c3943 000-2978b-aa0e-2e323931@192.168.
C-68 SIP IP Phone to SI P IP Phone: Cal l Retu rn Header: User-Agent: Ci sco IP Phone/ Rev. 1/ SIP enable d Header: Accept: applic ation/sdp Header: Contact: sip:6 711@192.
SIP IP Phone to SIP I P Phone: Call Return C-69 -------------------------------------- --------------------------- sip-req: ACK sip:*69@192.168.26.200 :5060;user=phone SIP/2.0 [192.16 8.26.180:5060- >192.168.26.200:5060] Header: Via: SIP/2.0/U DP 192.
C-70 SIP IP Phone to SI P IP Phone: Cal l Retu rn Header: To: <sip:*69@1 92.168.26.180:5060> Header: Call-ID: c2943 000-2e262-b27e-2e323931@192.168.
SIP IP Phone to SIP I P Phone: Call Return C-71 sip-req: INVITE sip:6715@192.168.26 .200:5060;user=phone SIP/2.0 [19 2.168.26.180:5060- >192.168.26.200:5060] Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=1 Header: Via: SIP/2.0/U DP 192.168.26.
C-72 SIP IP Phone to SI P IP Phone: Cal l Retu rn -------------------------------------- --------------------------- Header: v=0 Header: o=CiscoSy stemsSIP-IPPhone-UserAgent 8962 2 811 IN IP4 192.168.26.10 Header: s=SIP Call Header: c=IN IP4 192.1 68.
SIP IP Phone to SIP I P Phone: Call Return C-73 Header: m=audio 29956 RTP/AVP 0 101 Header: a=rtpmap:0 pcm u/8000 Header: a=rtpmap:101 t elephone-event/8000 Header: a=fmtp:101 0-1 1 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.
C-74 SIP IP Phone to SI P IP Phone: Cal l Retu rn Header: Route: <sip:67 15@192.168.26.180:5060;maddr=192 .168.26.180>, <sip:6711@192.168.26.10:5060> Header: Content-Length : 0 -----------.
User Agent to User Agent: Call W aiting C-75 User Agent to User Agent: Call W aiting Call Scenario Figur e C- 25 illu strates t he foll owing cal l scenari o: • User A cal ls User B • While Users A and B are in conversation, User C call s User A • User A is notified that another cal ler attempting to connec t Figure C-25.
C-76 User Agent t o User Agen t: Call Wai ting Figure C-26. User Agent to User Agent : Call Wa iting — Diagr am 1 VOCAL User Agent UA Marshal 1. INVITE 2. 100 Redirect Server 3. INVITE 4. 302 5. ACK 6. INVITE 7. 302 8. ACK VOCAL User Agent 9. INVITE 10.
C-77 User Agent t o User Agen t: Call Wai ting Figure C-27. User Agent to User Agent : Call Wa iting — Diagr am 2 VOCAL User Agent UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 16. INVITE 17. 100 18. INVITE 19. 302 20. ACK 21. INVITE 22.
C-78 User Agent t o User Agen t: Call Wai ting Figure C-28. User Agent to User Agent : Call Wa iting — Diagr am 3 VOCAL User Agent UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 32. INVITE 33. 200 34. INVITE 35. 200 36. ACK 37. ACK 38.
User Agent to User Agent: Call W aiting C-79 Call T r ace The foll owing call trac e shows a thir d par ty attempting t o connect to a phone that is engaged in conv ersation wit h another phone.
C-80 User Agent t o User Agent: Call Wai ting Header: F -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 302 Moved Temporar ily [192.168.
User Agent to User Agent: Call W aiting C-81 -------------------------------------- --------------------------- sip-req: ACK sip:5220@192.168.66.20 0:5060;user=phone SIP/2.0 [192.1 68.66.180:5060- >192.168.66.200:5060] Header: Via: SIP/2.0/U DP 192.
C-82 User Agent t o User Agent: Call Wai ting Header: From: UserAgen t<sip:5221@192.168.66.1:5060;use r=phone> Header: To: 5220<sip:5 220@192.168.66.180:5060;user=pho ne> Header: Call-ID: 4732a 6465cfdffbdc0d38708c0728708@192. 168.66.1 Header: CSeq: 1 INVITE Header: Contact: <sip: 5220@192.
User Agent to User Agent: Call W aiting C-83 Header: From: UserAgen t<sip:5221@192.168.66.1:5060;use r=phone> Header: To: 5220<sip:5 220@192.168.
C-84 User Agent t o User Agent: Call Wai ting Header: m=audio 60335 RTP/AVP 0 Header: a=rtpmap:0 PCM U/8000 Header: a=ptime:20 Header: * -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.
User Agent to User Agent: Call W aiting C-85 Header: Content-Length : 0 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: ACK sip:5220@192.168.66.
C-86 User Agent t o User Agent: Call Wai ting Header: Route: <sip:5220@192.168.66.180:5060;maddr=19 2.168.66.180>,<sip:5220@192.168. 66.180:5060;maddr=1 92.
User Agent to User Agent: Call W aiting C-87 sip-res: SIP/2.0 200 OK [192.168.66. 180:5060- >192.168.66.3:5060] Header: Via: SIP/2.0/U DP 192.168.66.3:5060 Header: From: UserAgen t<sip:5222@192.168.66.3:5060;use r=phone> Header: To: 5220<sip:5 220@192.
C-88 User Agent t o User Agent: Call Wai ting SDP Headers -------------------------------------- --------------------------- Header: v=0 Header: o=- 1573383876 1573383876 IN IP4 192.
User Agent to User Agent: Call W aiting C-89 Header: Content-Type: application/sdp Header: Content-Type: application/sdp Header: Content-Length : 168 Header: Content-Length : 168 ---------------------.
C-90 User Agent t o User Agent: Call Wai ting sip-req: BYE sip:5220@192.168.66.18 0:5060;maddr=192.168.66.180 SIP/ 2.0 [192.168.66.2:5060->192.168.66.180:506 0] Header: Via: SIP/2.0/U DP 192.168.66.2:5060 Header: From: 5220<sip :5220@192.168.66.
SIP IP Phone to SIP IP Phone: Forward to V oice Mail C-91 SIP IP Phone to SIP IP Phon e: Forward to V oice Mail Call Scenario Figur e C- 29 illu strates t he foll owing cal l scenari o: • User A cal ls User B • User B does not answer the call • The call is for warded to the voi ce mail featu re ser ver Figure C-29 .
C-92 SIP IP Phone to SIP IP Phone: Forward to V oi ce Mail Figure C-30 . SIP IP Phone to SIP IP Pho ne: Forward to V oice Mail — Di agram 1 SIP Phone UA Marshal 1. INVITE 2. 100 Redirect Server 3. INVITE 4. 302 5. ACK Feature Server: FNA 6. INVITE 7.
C-93 SIP IP Phone to SIP IP Phone: Forward to V oi ce Mail Figure C-31 . SIP IP Phone to SIP IP Pho ne: Forward to V oice Mail — Di agram 2 SIP Phone UA Marshal Redirect Server Feature Server: FNA SIP Phone 16. 180 17. 180 18. 180 19. CANCEL 20. 200 21.
C-94 SIP IP Phone to SIP IP Phone: Forward to V oi ce Mail Figure C-32 . SIP IP Phone to SIP IP Pho ne: Forward to V oice Mail — Di agram 3 SIP Phone UA Marshal Redirect Server Feature Server: FNA SIP Phone 32.
SIP IP Phone to SIP IP Phone: Forward to V oice Mail C-95 Call T r ace The followi ng call trace shows a SIP IP phone attempting to cal l another on- network SI P IP phon e. The second phone i s unanswere d and the c all is re- initiate d with the V oice Mail serv er .
C-96 SIP IP Phone to SIP IP Phone: For ward to V oice Mai l Header: t=0 0 Header: m=audio 23994 RTP/AVP 0 101 Header: a=rtpmap:0 pcm u/8000 Header: a=rtpmap:101 t elephone-event/8000 Header: a=fmtp:10.
SIP IP Phone to SIP IP Phone: Forward to V oice Mail C-97 SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:5210@192.168.56 .180:5060 SIP/2.0 [192.168.56. 220:5074- >192.168.56.180:5060] Header: Via: SIP/2.
C-98 SIP IP Phone to SIP IP Phone: For ward to V oice Mai l -------------------------------------- --------------------------- Header: v=0 Header: o=CiscoSystems SIP-IPPhone-UserAgent 25678 2814 0 IN IP4 192.168.10.18 Header: s=SIP Call Header: c=IN IP4 192.
SIP IP Phone to SIP IP Phone: Forward to V oice Mail C-99 Header: a=rtpmap:0 pcm u/8000 Header: a=rtpmap:101 t elephone-event/8000 Header: a=fmtp:101 0-1 1 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.
C-100 SIP IP Phone to SIP IP Phone: For ward to V oice Mai l Header: Via: SIP/2.0/U DP 192.168.56.220:5074;branch=10 2 Header: From: <sip:521 8@192.168.10.18:5060> Header: To: <sip:5210@ 192.168.56.180:5060> Header: Call-ID: c2943 000-482e7-61caa-2e323931@192.
SIP IP Phone to SIP IP Phone: Forward to V oice Mail C-101 Header: Contact: <sip: 6500@192.168.56.220:5082> Header: Record-Route: <sip:192.168.56.220:5060;maddr=192.168 .56.220>,<sip:5210@192.168.56.22 0:5074;maddr=192.16 8.56.220>,<sip:5210@192.
C-102 SIP IP Phone to SIP IP Phone: For ward to V oice Mai l Header: Route: <sip:192.168.56.220:5060;maddr=192.168 .56.220>,<sip:6500@192.168.
User Agent to User Agent: Consulted T r ansfer C-103 User Agent to User Agent: Consulted T ransfer Call Scenario Figur e C- 33 illu strates t he foll owing cal l scenari o: • User A cal ls User B. • User B puts User B on hold and notif ies User C about User A ’ s c all.
C-104 User Agent t o User Agent: Consulted Transf er Figure C-34. User Agent to User Agent : Consulted T r ansfer — Diagram 1 VOCAL User Agent UA Marshal 1. INVITE 2. 100 VOCAL User Agent 3. INVITE 4. 200 5. 200 6. ACK 7. ACK 8. INVITE 9. 100 10. INVITE 11.
C-105 User Agent t o User Agent: Consulted Transf er Figure C-35. User Agent to User Agent : Consulted T r ansfer — Diagram 2 VOCAL User Agent UA Marshal VOCAL User Agent VOCAL User Agent 16. 100 17. INVITE 18. 180 19. 180 20. 200 21. 200 22. ACK 23.
C-106 User Agent t o User Agent: Consulted Transf er Figure C-36. User Agent to User Agent : Consulted T r ansfer — Diagram 3 VOCAL User Agent UA Marshal VOCAL User Agent VOCAL User Agent 32. 100 33. TRANSFER 34. 100 35. INVITE 36. 100 37. INVITE 38.
C-107 User Agent t o User Agent: Consulted Transf er Figure C-37. User Agent to User Agent : Consulted T r ansfer — Diagram 4 VOCAL User Agent UA Marshal VOCAL User Agent VOCAL User Agent 48.
C-108 User Agent t o User Agent: Consulte d T ransfer Call T r ace The followi ng call trace shows a consulted cal l transfer between two SIP IP phones. -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:5221@192.
User Agent to User Agent: Consulted T r ansfer C-109 Header: -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 200 OK [192.168.66 .2:5060->192.
C-1 10 User Agent t o User Agent: Consulte d T ransfer SIP Headers -------------------------------------- --------------------------- sip-req: ACK sip:5221@192.168.66.2: 5060 SIP/2.0 [192.168.26. 180:5060- >192.168.66.2:5060] Header: Via: SIP/2.0/U DP 192.
User Agent to User Agent: Consulted T r ansfer C-1 1 1 Header: c=IN IP4 0.0.0 .0 Header: t=3174939344 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCM U/8000 Header: a=ptime:20 Header: j Ë --.
C-1 12 User Agent t o User Agent: Consulte d T ransfer -------------------------------------- --------------------------- sip-req: ACK sip:5221@192.168.66.2: 5060 SIP/2.0 [192.168.26. 180:5060- >192.168.66.2:5060] Header: Via: SIP/2.0/U DP 192.168.
User Agent to User Agent: Consulted T r ansfer C-1 13 Header: o=- 1113249245 1113249245 IN IP4 192.168.66.1 Header: s=VOVIDA Sessi on Header: c=IN IP4 192.
C-1 14 User Agent t o User Agent: Consulte d T ransfer Header: Record-Route: <sip:5222@192.168.26.180:5060;maddr=19 2.168.26.180>,<sip:5222@192.
User Agent to User Agent: Consulted T r ansfer C-1 15 SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 100 Trying [192.16 8.26.180:5060->192.168.66.1:5060 ] Header: Via: SIP/2.0/U DP 192.168.66.1:5060 Header: From: UserAgen t<sip:5220@192.
C-1 16 User Agent t o User Agent: Consulte d T ransfer Header: Contact: <sip: 5222@192.168.66.3:5060;user=phon e> Header: Content-Type: application/sdp Header: Content-Length : 166 -------------.
User Agent to User Agent: Consulted T r ansfer C-1 17 sip-req: TRANSFER sip:5221@192.168. 66.2:5060 SIP/2.0 [192.168.26. 180:5060- >192.168.66.2:5060] Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.
C-1 18 User Agent t o User Agent: Consulte d T ransfer Header: CSeq: 1 INVITE Header: Subject: Vovid aINVITE Header: Record-Route: <sip:5222@192.168.26.180:5060;maddr=19 2.168.26.180>,<sip:5222@192.168. 26.180:5060;maddr=1 92.168.26.180> Header: Contact: <sip: 5221@192.
User Agent to User Agent: Consulted T r ansfer C-1 19 Header: s=VOVIDA Sessi on Header: c=IN IP4 192.1 68.66.3 Header: t=3174939395 0 Header: m=audio 23466 RTP/AVP 0 Header: a=rtpmap:0 PCM U/8000 Head.
C-120 User Agent t o User Agent: Consulte d T ransfer -------------------------------------- --------------------------- sip-req: BYE sip:5222@192.168.26.18 0:5060;maddr=192.168.26.180 SIP/ 2.0 [192.168.66.1:5060->192.168.26.180:506 0] Header: Via: SIP/2.
User Agent to User Agent: Consulted T r ansfer C-121 Header: Content-Length : 0 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 200 OK [192.
C-122 User Agent t o User Agent: Blind T ransfer User Agent to User Agent: Blind T ransfer Call Scenario Figur e C- 38 illu strates a call sc enario in which: • User A cal ls User B. • User B transfer s the cal l to User C without consulting User C.
C-123 User Agent t o User Agent: Blind Transf er Figure C-39. User Agent to User Agent : Blind T ran sfer — Diagram 1 VOCAL User Agent UA Marshal 1. INVITE 2. 100 VOCAL User Agent 3. INVITE 4. 180 5. 180 6. 200 7. 200 8. ACK 9. ACK 10. INVITE 11. 100 12.
C-124 User Agent t o User Agent: Blind Transf er Figure C-40. User Agent to User Agent : Blind T ran sfer — Diagram 2 VOCAL User Agent UA Marshal VOCAL User Agent VOCAL User Agent 16. ACK 17. INVITE 18. 100 19. INVITE 20. 180 21. 180 22. INVITE 23. 100 24.
C-125 User Agent t o User Agent: Blind Transf er Figure C-41. User Agent to User Agent : Blind T ran sfer — Diagram 3 VOCAL User Agent UA Marshal VOCAL User Agent VOCAL User Agent 32. 100 33. INVITE 34. 180 35. 180 36. 200 37. 200 38. BYE 39. CANCEL 40.
C-126 User Agent t o User Agent: Blind Transf er Figure C-42. User Agent to User Agent : Blind T ran sfer — Diagram 4 VOCAL User Agent UA Marshal VOCAL User Agent VOCAL User Agent 48.
User Agent t o User Agent: Blind Transf er C-127 Call T r ace The followi ng call trace shows an unconsul t ed call transf er . -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:5221@192.
C-128 User Agent t o User Agent: Blind T ransfer -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 180 Ringing [192.1 68.66.2:5060->192.
User Agent t o User Agent: Blind Transf er C-129 Header: s=VOVIDA Sessi on Header: c=IN IP4 192.1 68.66.2 Header: t=3174939460 0 Header: m=audio 23466 RTP/AVP 0 Header: a=rtpmap:0 PCM U/8000 Header: a.
C-130 User Agent t o User Agent: Blind T ransfer -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:5221@192.168.66 .2:5060 SIP/2.0 [192.
User Agent t o User Agent: Blind Transf er C-131 Header: c=IN IP4 192.1 68.66.2 Header: t=3174939460 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCM U/8000 Header: a=ptime:20 Header: ‚ ----.
C-132 User Agent t o User Agent: Blind T ransfer -------------------------------------- --------------------------- sip-req: INVITE sip:5222@192.168.66 .3:5060 SIP/2.0 [192.168.26. 180:5060- >192.168.66.3:5060] Header: Via: SIP/2.0/U DP 192.168.26.
User Agent t o User Agent: Blind Transf er C-133 Header: s=VOVIDA Sessi on Header: c=IN IP4 0.0.0 .0 Header: t=3174939482 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCM U/8000 Header: a=ptim.
C-134 User Agent t o User Agent: Blind T ransfer -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: TRANSFER sip:5221@192.168. 26.180:5060;maddr=192.
User Agent t o User Agent: Blind Transf er C-135 Header: o=- 1986829226 1986829226 IN IP4 192.168.66.2 Header: s=VOVIDA Sessi on Header: c=IN IP4 192.1 68.
C-136 User Agent t o User Agent: Blind T ransfer -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 200 OK [192.168.66 .2:5060->192.168.
User Agent t o User Agent: Blind Transf er C-137 SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 200 OK [192.168.66 .2:5060->192.168.26.180:5060] Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.
C-138 User Agent t o User Agent: Blind T ransfer Header: ÷Ï -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.0 200 OK [192.168.26 .180:5060->192.
User Agent t o User Agent: Blind Transf er C-139 sip-req: BYE sip:5222@192.168.66.3: 5060 SIP/2.0 [192.168.26. 180:5060- >192.168.66.3:5060] Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.
C-140 JT API JT API Call Scenario Figur e C- 43 illu strates t he foll owing cal l scenari o: • A user use s a JT API User Agent on a PC t o remotely instruct SIP Pho ne A to call SIP Phone B. Figure C-43. JT API Call Flow Diagram Fi gur es C-44, C-45, C-46 and C-47 show thir d p arty call cont rol through a JT API se rver .
C-141 JT API Figure C-44. Call Flow Diagram: JT API — Diagr am 1 JTAPI User Agent Redirect Server 1. INVITE 2. 302 3. ACK UA Marshal 4. INVITE 5. 100 6.
C-142 JT API Figure C-45. Call Flow Diagram: JT API — Flow Diagr am 2 JTAPI User Agent Redirect Server UA Marshal VOCAL User Agent SIP Phone 16. 302 17. 200 18. 200 19. ACK 20. ACK 21. 100 22. 100 23. INVITE 24. 302 25. INVITE 26. 302 27. ACK 28. INVITE 29.
C-143 JT API Figure C-46. Call Flow Diagram: JT API — Flow Diagr am 3 JTAPI User Agent Redirect Server UA Marshal VOCAL User Agent SIP Phone 32. 100 33. 180 34. 180 35. 180 36. 180 37. 200 38. 200 39. BYE 40. BYE 41. 200 42. 200 43. 200 44. 200 45. 200 46.
C-144 JT API Figure C-47. Call Flow Diagram: JT API — Flow Diagr am 4 JTAPI User Agent Redirect Server UA Marshal VOCAL User Agent SIP Phone 48. ACK 49.
JT API C-145 Call T r ace The followi ng call trace shows thir d part y call control through a JT API server . -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:6710@192.
C-146 JT API Header: Content-Length : 0 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:6710@192.168.26 .200:5060;user=phone SIP/2.0 [19 2.
JT API C-147 Header: Via: SIP/2.0/U DP 192.168.5.11:25060;branch=301 Header: From: <sip:-@1 92.168.5.11:25060> Header: To: 6710<sip:6 710@192.
C-148 JT API SDP Headers -------------------------------------- --------------------------- Header: v=0 Header: o=- 1735072859 1735072859 IN IP4 192.168.
JT API C-149 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCM U/8000 Header: a=ptime:20 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.
C-150 JT API Header: Call-ID: 10878 93762978930@192.168.5.11 Header: CSeq: 2 TRANSF ER Header: Content-Length : 0 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:6711@192.
JT API C-151 Header: Via: SIP/2.0/U DP 192.168.5.11:15060;branch=202 Header: From: UserAgen t<sip:6710@192.168.22.36:5060> Header: To: 6711<sip:6 711@192.168.26.200:5060;user=pho ne> Header: Call-ID: 45889 8268105186@192.168.5.11 Header: CSeq: 1 INVITE Header: Contact: <sip: 6711@192.
C-152 JT API sip-req: ACK sip:6711@192.168.26.20 0:5060;user=phone SIP/2.0 [192.1 68.26.180:5060- >192.168.26.200:5060] Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=3 Header: From: UserAgen t<sip:6710@192.168.22.36:5060> Header: To: 6711<sip:6 711@192.
JT API C-153 sip-res: SIP/2.0 200 OK [192.168.22 .36:5060->192.168.26.180:5060] Header: Via: SIP/2.0/U DP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/U DP 192.168.5.11:25060 Header: From: <sip:-@1 92.168.5.11:25060> Header: To: 6710<sip:6 710@192.
C-154 JT API Header: From: UserAgen t<sip:6710@192.168.22.36:5060> Header: To: 6711<sip:6711@192.168.26.200:5060;user =phone>;tag=c29430003e2620-0 Header: Call-ID: 45889 8268105186@192.168.5.11 Header: Server: Cisco IP Phone/ Rev. 1/ SIP enabled Header: Contact: sip:6 711@192.
JT API C-155 Header: o=CiscoSystems SIP-IPPhone-UserAgent 26487 2824 7 IN IP4 192.168.26.10 Header: s=SIP Call Header: c=IN IP4 192.1 68.26.10 Header: t=0 0 Header: m=audio 23206 RTP/AVP 0 101 Header:.
C-156 JT API Header: CSeq: 1 ACK Header: Route: <sip:67 11@192.168.26.10:5060> Header: Proxy-Authoriz ation: Basic vovidaClassXswitch Header: Content-Length : 0 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.
JT API C-157 Header: CSeq: 2 BYE Header: Route: <sip:js @192.168.5.11:15060> Header: Content-Length : 0 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: BYE sip:js@192.
C-158 JT API Header: From: 6711<sip:6711@192.168.26.200:5060;user =phone>;tag=c29430003e2620-0 Header: To: UserAgent< sip:6710@192.168.22.36:5060> Header: Call-ID: 45889 8268105186@192.168.5.11 Header: CSeq: 2 BYE Header: Contact: <sip: js@192.
Ad Hoc Conference Call Betw een User Agents C-159 Ad Hoc Conference Call Between Us er Agent s Call Scenario Figur e C- 48 illu strates t he foll owing cal l scenari o: • User A cal ls User B • User A brings User C into the con vers ation via ad hoc confere ncing.
C-160 Ad Hoc Conference Call Between User Agents Figure C-49. User Agent to User Agent to User Agent: Ad Hoc Conference Call: Disconnect Last Party — Diagram 1 VOCAL User Agent UA Marshal 1. INVITE 2. 100 Redirect Server 3. INVITE 4. 302 5. ACK 6. INVITE 7.
C-161 Ad Hoc Conference Call Between User Agents Figure C-50. User Agent to User Agent to User Agent: Ad Hoc Conference Call: Disconnect Last Party — Diagram 2 VOCAL User Agent UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 16. INVITE 17.
C-162 Ad Hoc Conference Call Between User Agents Figure C-51. User Agent to User Agent to User Agent: Ad Hoc Conference Call: Disconnect Last Party — Diagram 3 VOCAL User Agent UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 32. 180 33.
C-163 Ad Hoc Conference Call Between User Agents Figure C-52. User Agent to User Agent to User Agent: Ad Hoc Conference Call: Disconnect Last Party — Diagram 4 VOCAL User Agent UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 48. 100 49.
C-164 Ad Hoc Conference Call Between User Agents Figure C-53. User Agent to User Agent to User Agent: Ad Hoc Conference Call: Disconnect Last Party — Diagram 5 VOCAL User Agent UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 64. 200 65.
Ad Hoc Conference Call Betw een User Agents C-165 Call T r ace The followi ng call trace shows an ad hoc conference cal l bet ween thr ee users. -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:5203@192.
C-166 Ad Hoc Conference Call Between User Agents sip-res: SIP/2.0 302 Moved Temporar ily [192.168.46. 200:5060- >192.168.46.180:5060] Header: Via: SIP/2.0/U DP 192.168.46.180:5060;branch=1 Header: Via: SIP/2.0/U DP 192.168.46.1:5060 Header: From: seymour< sip:5201@192.
Ad Hoc Conference Call Betw een User Agents C-167 Header: From: seymour< sip:5201@192.168.46.1:5060;user= phone> Header: To: 5203<sip:5 203@192.
C-168 Ad Hoc Conference Call Between User Agents Header: Record-Route: <sip:5203@192.168.46.180:5060;maddr=19 2.168.46.180>,<sip:5203@192.168.
Ad Hoc Conference Call Betw een User Agents C-169 ====================================== =========================== First call establi shed -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:5203@192.
C-170 Ad Hoc Conference Call Between User Agents Header: Via: SIP/2.0/U DP 192.168.46.180:5060;branch=4 Header: Via: SIP/2.0/U DP 192.168.46.180:5060;branch=2 Header: Via: SIP/2.0/U DP 192.168.46.1:5060 Header: From: seymour< sip:5201@192.168.46.1:5060;user= phone> Header: To: 5203<sip:5 203@192.
Ad Hoc Conference Call Betw een User Agents C-171 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:5202@192.168.46 .180:5060;user=phone SIP/2.
C-172 Ad Hoc Conference Call Between User Agents Header: To: 5202<sip:5 202@192.168.46.180:5060;user=pho ne> Header: Call-ID: 64133 833318102@192.
Ad Hoc Conference Call Betw een User Agents C-173 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: INVITE sip:5202@192.168.46 .2:5060 SIP/2.0 [192.
C-174 Ad Hoc Conference Call Between User Agents SDP Headers -------------------------------------- --------------------------- Header: v=0 Header: o=- 979501686 979501686 IN IP4 192.
Ad Hoc Conference Call Betw een User Agents C-175 sip-req: INVITE sip:5202@192.168.46 .180:5060;maddr=192.168.46.180 S IP/2.0 [192.168.46.1:5060->192.168.46.180:506 0] Header: Via: SIP/2.0/U DP 192.168.46.1:5060 Header: From: seymour< sip:5201@192.
C-176 Ad Hoc Conference Call Between User Agents Header: Content-Length : 168 -------------------------------------- --------------------------- SDP Headers -------------------------------------- --------------------------- Header: v=0 Header: o=- 1554681096 1554681096 IN IP4 192.
Ad Hoc Conference Call Betw een User Agents C-177 Header: Via: SIP/2.0/U DP 192.168.46.1:5060 Header: From: seymour< sip:5201@192.168.46.1:5060;user= phone> Header: To: 5203<sip:5 203@192.168.46.180:5060;user=pho ne> Header: Transfer-To: < sip:818883831000@192.
C-178 Ad Hoc Conference Call Between User Agents Header: t=3177798665 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCM U/8000 Header: a=ptime:20 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-res: SIP/2.
Ad Hoc Conference Call Betw een User Agents C-179 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: TRANSFER sip:5203@192.168. 46.3:5060 SIP/2.0 [192.
C-180 Ad Hoc Conference Call Between User Agents -------------------------------------- --------------------------- Header: v=0 Header: o=CiscoSystems SIP-GW-UserAgent 1397 1625 IN IP 4 192.
Ad Hoc Conference Call Betw een User Agents C-181 sip-req: BYE sip:5203@192.168.46.3: 5060 SIP/2.0 [192.168.46. 180:5060- >192.168.46.3:5060] Header: Via: SIP/2.0/U DP 192.168.46.180:5060;branch=4 Header: Via: SIP/2.0/U DP 192.168.46.180:5060;branch=2 Header: Via: SIP/2.
C-182 Ad Hoc Conference Call Between User Agents Header: Via: SIP/2.0/U DP 192.168.46.180:5060;branch=4 Header: Via: SIP/2.0/U DP 192.168.46.180:5060;branch=2 Header: Via: SIP/2.0/U DP 192.168.46.1:5060 Header: From: seymour< sip:5201@192.168.46.1:5060;user= phone> Header: To: 5202<sip:5 202@192.
Ad Hoc Conference Call Betw een User Agents C-183 -------------------------------------- --------------------------- SIP Headers -------------------------------------- --------------------------- sip-req: BYE sip:5203@192.168.46.18 0:5060;maddr=192.168.
C-184 Ad Hoc Conference Call Between User Agents.
Symbols *69 call flow C-56 Numerics 1xx and 2xx B-3 3xx Responses B-3 4xx Responses B-3 5xx Responses B-4 6xx Responses B-4 900 # Admin Block 1-20 900 # User Block 1-23 A Access Level 1-4 Access list .
Index-2 Index (Cont inued) Long Distance Admin Bloc k 1-20 Long Distance User Blo ck 1-22 Marshal 1-19 Name 1-19 Password 1-21 Static Reg Enabled 1-21 Terminating Hos t 1-21 Terminati ng Port 1-21 Use.
Index-3 Index (Cont inued) R REGISTER B-2 Registrat ion C-3 Right mouse click menu 1-16 S Screen Edit User Administra tor view 1-11 User data view 1-31 Login 1 -3 SNMP messages 2-5 User Confi guration.
Index-4 Index (Cont inued) Users adding new users 1-10 deleting 1-2 7 edit multi ple users 1-28 finding user s 1-26 Load all users 1-24 – 1-27 procedure for edit ing 1-28 viewing indivi dually 1-16 .
An important point after buying a device Cisco Systems 1.3.0 (or even before the purchase) is to read its user manual. We should do this for several simple reasons:
If you have not bought Cisco Systems 1.3.0 yet, this is a good time to familiarize yourself with the basic data on the product. First of all view first pages of the manual, you can find above. You should find there the most important technical data Cisco Systems 1.3.0 - thus you can check whether the hardware meets your expectations. When delving into next pages of the user manual, Cisco Systems 1.3.0 you will learn all the available features of the product, as well as information on its operation. The information that you get Cisco Systems 1.3.0 will certainly help you make a decision on the purchase.
If you already are a holder of Cisco Systems 1.3.0, but have not read the manual yet, you should do it for the reasons described above. You will learn then if you properly used the available features, and whether you have not made any mistakes, which can shorten the lifetime Cisco Systems 1.3.0.
However, one of the most important roles played by the user manual is to help in solving problems with Cisco Systems 1.3.0. Almost always you will find there Troubleshooting, which are the most frequently occurring failures and malfunctions of the device Cisco Systems 1.3.0 along with tips on how to solve them. Even if you fail to solve the problem, the manual will show you a further procedure – contact to the customer service center or the nearest service center
