1. Press Ctrl + E to quickly enter the set connection system.
2. liststation: list all system extension numbers
3. changestation6720: Modify the extension number attribute, press F3 to save the configuration, Press F1 to save and exit or cancel the command.
4. displaystation6720: list all attributes of 6720
5. listtracestation6720, tracking ext. 6720
6. listtracetac4440, tracking relay Group
7. listvector: List call guides
8. listctilink: List CTI connections
9. displaycti-link1, listing CTI connections with sequence 1
10. removestation5004, delete ext. 5004
11. displaysystemcus: displays system authorization information.
12. changeoffsta6721, ec500, 18626208989, ars, 1
Changesta6721, ec500: enabled, ec500, autodial189999999999, call-park
Enable EC500 by clicking the left button on the AVAYA phone
Phone Resonance
TraceSM-xSIP session log
13. changetandem-calling-party-num
4, 60, 66806700, which indicates 4 characters in length, and 99 numbers starting with 60 characters are called 66806700 numbers.
14. changeuniform-dialplan0 dialing rules, 60, 4 represents the number between 6000-6099
15. changeappsanalysis0
A dialedstring is a networking route between internal nodes. It indicates the Starting number. If only one segment is restricted, enter a number such as 6001. If only one segment is restricted, enter 60, and set the length to 4, in this way, 99 numbers can be used. routepattern represents the type, and our 3 represents siptrunk.
16. After steps 13, 14, and 15 are completed, perform the following operations)
A. Enter systemmanager --> routing --> entitylinks --> new, configuration, sessionmanger, udp/tcp, port, trusted
B. Enter systemmanager --> routing --> sipentities --> new, configure name: asterisk, FQDN: 192.168.180.45sip server), SIPTimeerB/F: 4 by default), and add entitylinks
C. Enter systemmanager --> routing --> dialpatterns --> new, configure
Pattern: 60 (the start of the extension number. If it is set in ASA, it must correspond to the above). The length is 4 and an OriginatingLocationsandRoutingPolicies is added.
17. liconall lists all Board information
18. linodeall lists all host header information
19. statussignaling-group1 listing signaling service status
20. statustrunk1 lists the status of all trunk1 Channels
21. monitortraffictrunk-group real-time monitoring of relay group line occupation
22. The displaysyscdr list the interface for billing.
23. displayip-services: List cdr ports
24. stacdr list cdr status
25. Create a simple call center process (chasysfea confirms that 11th page EAS has been opened)
26. resetmedia-gateway restart g450
27. changetandem-calling-party-num
Example: 4,1011, 1, all, 66806412lol-pub
28. changeinc-call-handling-trmttrunk-group1 incoming number Conversion'
29. listreg lists the statuses of all registered phones
Call Center configuration process
A. Select a number as the Registry
B. addannouncementxxxx, type: integrated, group/board: 0 xaxx, unhook, press * 05 function code, enter the 4444ann number, press 1 to start recording, End recording press #, and then hook up, after a moment, you can call 4444 to hear the sound you just recorded.
C. addvdn entry number, destination: vectorNumber4
D. addvec4: Write Process
01wait-time2secshearingsilence
02collect1digitsafterannouncement4444fornone
03gotostep7ifdigits = 1
04gotostep10ifdigits = 2
05gotostep2ifdigits <> none
06
07queue-toskill150 (hunt-group) prih
08wait-time30secshearingmusic
09 stop
10queue-toskill150prih
11wait-time30secshearingmusic
12 stop
E. addhunt-group150, that is, adding a proxy group skill group), groupExtension: 4080 custom), acd \ queue \ vector changed to y, grouptype: ead-mia, skill: y
D. addagentloginID3000, turn to the second page, SN: 150 space 1, Agent login: Detach --> * 33 doodle + 3000 host, listagen check whether the login is successful, then, disconnect the host --> * 36. If you want to quit, disconnect the host --> * 34
F. Call 5003 to test the response to the 3000 agent.
AvayaauraMessaging6 configuration tutorialThe built-in email server is installed by default, which can be used only as an intranet email server to send emails to the Internet mailbox)
Log on to the MESSAGING Management Interface: Administration --> Messaging. The huntgroup in the premise CM has added a voicemail group similar to 4099, grouptype: ucd-mia, messagecenter: sip-adjunct, voicemailnumber \ voicemailhandle: 4099; and this 4099 must be included in listmediaanalysis, And the listroute-pattern must be: 99voicemail1100 and so on)
1. Set the gateway of the receiving server: Configure ServerSettings (Storage) --> ExternalHosts --> Add: IPAddress is the IP address of the receiving server, HOSTNAME: custom, Alias: alias: if the recipient has different receiving servers, enter them in IMAP/SMTPSettings (Storage) --> MailOptions --> MailboxGatewayMachineName is set to null, and ServerAlias is set to the alias you just set)
2. Set resolution for the receiving server: Enter the Message Server through SSH and modify the hosts correspondence: 0.0.0.0bt7086.commailbt (that is, the suffix after the recipient address @ corresponding IP address of the receiving server, but it may also be that the FQDN-HOSTNAME of the receiving server corresponds to the previous step)
3. Create a voicemail User: MessageingSystem (Storage) --> UserManagement --> Addauser --> set parameters: Firstname, Lastname, Displayname, MailboxNumber, Extension, password, Usermustchangevoicemessagingpasswordatnextlogin (example: 51, 17, 5117, 5117, 5117, 13579)
4. Set the user's corresponding voice email address: MessageingSystem (Storage) --> UserManagement --> enter 5003 users and click edit --> UserPreferences --> notify me --> email notification: check the following two check boxes to send an email and add the recording to it. Fill in the user's requested email address and save it.
5. test whether the call is successful: After making 3-4 calls, for example, 5003, enter the voice mailbox. After prompting that the recording is complete, check whether the mailbox has received the message and whether there is a message on the phone. If the email cannot be received, go to logs --> IMAP/SMTPMessaging --> Selectlogtoview --> SMTPInLog to check whether the gateway is sent to the receiving server, and SMTPOutLog to check whether the receiving Server Gateway is successfully sent.
AvayaauraSystemManager6.2(Common configurations and one-xcommunicator users added after integration with the presence Server)
Log on to the smgr Management page.
I. common functions and configurations
1. Check the data synchronization between smgr and the server of each role: Services --> Replication. Common roles: presenceserver and sessionmanager. If "Synchronized" is displayed, the synchronization is successful, if the role server is red and has been started, you can select the connection to be repaired and click "Repair". smgr will actively synchronize with the role server and refresh the server to see the real-time status!
2. Check whether the status of each link connected with sip is normal: Elements --> SessionManager --> If the Dashboard is green and the words "Hook" and "up" are correct, if you want to carefully check whether systemstatus --> sipentitymonitoring is green, green indicates OK. Generally, common roles are: links to clan boards, conferencebridge links, and Presence links, as long as the up status is displayed, it is OK.
3. Integration with role servers: Elements --> Routing
A. Domains: Enter the customer's domain Name, for example, Name: ceibs.edu.
B. Locations: The identifier of the server. Fill in the following fields: Name (custom) and LocationPattern, for example, ceibs-asm and 172.16.8 .*
C. SIPEntities: the entity that the SIP entity connects to. Generally, there are sessionmanager, CM, conference, and presence. For example, name (presence) and FQDNorIPAddress (ps address), Type (other), Location (ceibs-asm), TimeZone (shanghai), SIPLinkMonitoring (enabled), EntityLinks (ceibs-asm \ tls \ 5061 \ presence \ 5061 \ trusted) for example, ceibs-asm \ tcp & udp \ 5060 \ conference \ 5060 \ trusted), and cm is also tls
D, EntityLinks, commonly used: asm-ps \ asm \ tls \ 5061 \ ps \ 5061 \ trusted; s8800-cm \ asm \ tls \ 5061 \ cmalias \ 5061 \ trusted; sm-aac-tcp \ asm \ tcp \ 5060 \ conference \ 5060 \ trusted; sm-aac-udp \ asm \ udp \ 5060 \ conference \ 5060 \ trusted
4. Create a presence User: User --> UserManagement --> ManagerUser --> new, identity: LastName \ FirstName \ LoginName \ AuthenticationType \ Password \ Language \ TimeZone \: 7002 \ 7002 \ 7002@ceibs.edu \ basic \ Avaya123 (password security level is too high) \ chinese \ beijing;
Communicationprofile: communicationprofilepassword \ communicationaddress \ Sessionmanagerprofile \ primarysessionmanager \ originationapplicationsequence \ homelocation \ cmendpointprofile \ system \ profiletype \ extension \ template \ port
: 123456 \ avayasip-7002-ceibs \ ceibs-asm \ cm1 \ ceibs-asm \ After the input extension will automatically fill up
After entering 7002, click "editendpoint" and select "template --> 9630sip". The fields to be filled in are automatically displayed. Modify classofrestriction (COR): 2, modify FeatureOptions --> Features --> select IPsoftPhone.
Add a contact. Otherwise, the contact state cannot be seen during the test: contacts --> add. After the selection is complete, set the "presencebuddy" field to yes; otherwise, the contact state information cannot be seen.
Finally, submit the test to check whether the test is successful.
If the synchronization is successful, siptrunk is normal. Check whether private-numbering has a corresponding number segment in CM.
Presence Server Installation tutorial
Install rhel5.7 --> install software
1. Install rhel5.7. Configure an IP address and dns, remove all partitions and format them. The simplest installation is to install basesystem only, in system-tools, select watchdog and x-windows for graphical installation), set the time zone after restart, ntp (if any ), disable the firewall and selinux and set its startup level to 12345 off. Go to the CD and install the rpm-ivhlibtool-ltdl plug-in. Set the hosts to no longer use if DNS exists. In reslov. set dns in conf). For smgr and asm, download jdk1.6.0 _ 18 and install it under the/usr/java directory ,. /*. bin decompress the package and set the Java environment. Enter JAVA_HOME =/usr/java/jdk1.6.0 _ 18 at the beginning and end of vi/etc/profile and press enter path = $ JAVA_HOME/bin /: $ PATH: Press ENTER exportPATHJAVA_HOME and press ENTER chmod-Ra + rx $ JAVA_HOME. Save the settings and save the settings as source/etc/profile. Then, run the command in vi/etc/profile. d/java. sh, enter the same content and take effect, test whether java-version is OK
2. Copy the pssoftware about MB to the/tmp directory and install xming to run it as long as the value is 0. Open putty and select connection --> SSH --> X11 --> EnableX11forwarding, enter localhost: 0 to run and enter tmp. /PS -*. sh-ci: If a graphical interface is displayed, it indicates OK.
3. At the beginning, the system will verify that the memory is not enough. If it is not enough, it is only the installation time. Generally, 32 GB will be faster. The installation process is about 5-10 minutes. If it is half done, it will take at least 1-2 hours.
4. precautions during installation: the installation mode is standard, and SMGR prompts you to enter the FQDN, user name, and password of smgr. The password is the password for smgr to log on to the web interface ); PS configure routerservicename and smgr --> elements --> presence --> configuration --> To's @. Otherwise, communication fails. The system prompts that you do not need To select multiple components by default; sessionmanageraddress: Enter the second address of sm, that is, the address used by the sip Phone for login. TrustManagementServiceConfiguration the Enrollment password is smgr --> services --> security --> certificates --> EnrollmentPassword
5. After the installation is complete, go to the web page and check whether the services are available.
How to Use the VSP PlatformGenerally, the role server is in the virtual machine)
1. Upload licenses: log onto the VSPweb page --> ServerManagement --> LicenseManagement --> LaunchWebLMLicenseManager
Virtual Machine Management
1. view the virtual machine information xmlist
2. view the VM tool xmhelp
3. Restart the VM xmreboothostname
Integration of outlook2007 and AAC
Prerequisites: You have installed outlook2007. It is best to install it all! The XP Client is used for testing.
1. When OutlookConferenceSchedulerClientSetup2007.msi is installed on the client, the system will prompt you to enter the bridge IP address, wportal user name and email user) and password.
2. Install the Microsoft Plug-In Office2007PIA on the client. There may be no prompts, but you don't need to worry about it. It's actually done! At this point, there are basically no settings on the client!
3. Enable outlook reservation on the server: Enter SMGR (connected to AAC) --> Elements --> MeetingExchange --> ClientRegistration --> General --> UCIntegrationClients --> UCIntegrationSettings --> Schedule --> off/hide ==> on, the client can Schedule a meeting to see the effect.
4. Set the time zone: If this parameter is not set, the default plug-in is the utc field in the read database. The default value is one day. We can only schedule the meeting on the next day:
A. enter SMGR (connected to AAC) --> Elements --> MeetingExchange --> ClientRegistration --> Timezone --> select CRS --> NEW to a time zone --> note that GMT in China takes 8 hours, that is, the offset value is set to 480, other parameters are not important. Save and apply the changes; otherwise, the changes will not take effect.
B. go to SMGR (connected to AAC) --> Elements --> MeetingExchange --> ClientRegistration --> General --> GeneralSettings --> SiteTimezone (set to the China time zone just created ), save and apply the changes, but not here!
C. Enter the "user and password of the corresponding client" in the webportal: My account --> preferred time zone --> the new time zone
D. Test whether it is normal.
AAC Configuration
1. Set the proportion of the instant meeting: The SMGR has been connected to AAC) -->/Elements/MeetingExchange/AudioConferencing/ConferenceFeatures --> select the corresponding bridge --> conferencesetures --> On-DemandPct %. Set this parameter to 20%.
2. Instant meeting trial: Go to CRS --> createreservation --> duration (duration) and participation (number of meeting persons)
3. Add the access number: Enter the SMGR that has been connected to AAC) --> Elements --> Routing --> RoutingPolicies --> DialPatterns)
4. Change the access number: Enter the SMGR already connected to AAC) --> Elements/Routing/RoutingPolicies --> TO-AAC (depending on the situation) --> addDialPatterns (that is, add a new access number ), after that, you also need to add routes and huntgroup in CM. Otherwise, the IP phone cannot be connected. After completing the last step, you must go to "Moreactions --> reflashalldata" in RoutingPolicies; otherwise, the added number will not take effect.
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