CINEMA-GUI: User Manual

Table of contents


Overview

Cinema-GUI allows configuration of various user and system profiles from the web using a browser. It is a front end for the database tables used by the system. There are two types of users: regular user or administrator. An administrator is identified by some additional privileges over a regular user. The first user created becomes the administrator. He can add additional users as administrator later, or change the status from administrator to regular user or vice-versa. A regular user can not access profiles of other user or change status of himself or other users.

Users can login from the web page by providing their userid and password. Userid is a unique identifier for a user and is of the form user@domain.

User accounts and various information are maintained in SQL database.

The system administrator must install the system before the users can use it. See the installation manual available with distribution.

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Creating a user

A new user is created by visiting the main web page, say, http://www.example.com/cgi-bin/cinema. (*NOTE* the followed link in this example is for our installation. Your installation will have a different URL). Enter the userid of the form user@domain as shown below. In the following screenshot user ka191@cs.columbia.edu is being created. Your domain may be different from "cs.columbia.edu". You do not need to change the Realm edit box. Note that the userid MUST be a valid email address. An email notification is sent to the userid informing about the password. When a new user is created the default user profile are borrowed from the that of user "default@domain". For example in this case the user profile is copied from "default@cs.columbia.edu". The user can change his profile later. The default profile can be altered by the administrator, typpically during installation. The initial password is randomly choosen. Users are strongly encouraged to change the password immediately after creation, by logging into the system.

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Logging in

Visit the same page again and enter your userid as "user@domain". Click on Login or add user button. If the user is already created then the system will prompt for the password. After logging in you will get the user menu frame similar to the following diagram.
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The exact number of menu items depends on your user type: administrtor or regular user, and your installation type: whether you have voice mail or conferencing.

Make sure that your browser is configured to enable Java and JavaScript. For example, for Netscape, go to Edit->Preferences->Advanced, and click the check boxes for "Enable Java" and "Enable JavaScript". Also it important to enable "Accept all cookies". Your browser should support frames. Both Netscape and Internet Explorer support frames.

Placing your mouse on the appropriate icon will display the purpose of that icon. various icons correspond to "Edit user profile", "Edit aliases", "Edit contacts", "Log out", and so on. Administrators also have additional menu items for "List users", "Add user", "Gateway class", and "Configure".

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Edit User profile

After you click on Edit user icon you can edit various user profile information.
Realm
This is the realm or the prompt for asking for your password. Normally you should keep it the same as your domain name, e.g., cs.columbia.edu. To change the realm you MUST supply the password. The reason is that the password is stored as encrypted hash of the user id, realm and password. Changing the realm will make the encrypted hash invalid. So you must also fill the "New password" and the "Reenter password for verification" edit boxes.
New password
You need to supply the new password and reenter it for verification only if you wish to change your password or your realm. It is strongly encouraged to change the password frequently and not to use common or dictionary words.
Authentication
The SIP server supports authentication based on basic or digest authentication algorithms. The authentication mechanism is a global configuration which can be changed by the administrator. However, when authentication is needed can be set on per-user basis. The global configuration option overrides per-user configuration. Global configuration can set the authentication to "digest", "basic" or "none". If the global configuration is set to use authentication then all SIP REGISTER requests must be authenticated. For INVITE and other requests (used for making and terminating the calls), the per user configuration can be set to (1) never: for no proxy-authentication for making a call, (2) request: try requesting authentication but complete the call even if authentication fails, or (3) required: always requires proxy authentication. For initial system users can set this value to "never" to allow anobody to reach them.

Note that the registration authentication is a system wide global configuration option and can not be changed on per-user basis, unlike call authentication.

Most User set the algorithm to MD5 for authentication.

voice mail
Voice mail is an optional component. Ask your administrator to license sipum to avail the voicemail facility.

If the user has been configured to have voice mail then he can set the RTSP URL for the outgoing message prompt which gets played when the user's phone is busy or there is no response. Alternatively you may upload your own recorded message as the prompt. This prompt is played when your phone is busy or there is no response.

When a new voice mail is received an email notification is sent to the user. The format of the email can be altered on a per-user basis. For example, a user using web based email system would prefer to have HTTP URLs in the email.

Users can also set the timeout after which the call is forwarded to the voicemail system. And the maximum size of each message allowed for this user. The total size of all the messages is a system level global configuration option.

This page also allows you to sign-out from the service. Another confirmation dialog is presented before actually deleting user account.

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Edit Aliases

After you click on "Aliases" icon , you can change your aliases. An user can have multiple aliases. These aliases are alternate URLs for you. For example is the userid is bob@example.com and an alias adde for this user is Bobby@example.com, then any call for sip:Bobby@example.com will be forwarded to bob@example.com.

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Edit Contacts

Contacts page allows you to view and edit current SIP registrations. This is also useful for follow-me type of service where you can enter a contact as "sip:bob.wilson@california-hotel.com", and all the calls to you will be forwarded to this new address.

The contact must be a valid URL. Usually the SIP server does not forward the call to a non-sip URL.

Preference for a particular contact has value between 0.0 to 1.0. Higher value is more preferred. So 1.0 is the most preferred contact. The SIP server in proxy mode tries to contact the locations based on the preference values, higher preference first. All contacts with same preference value are contacted at the same time, i.e., all those phones will ring simultaneously.

Users can specify the expiry time. If nothing is specified then the contact location never expires.

Users can choose whether the contact locations should be contacted in proxy or redirect mode. Proxy mode is transparent to the calling party and hence may be preferred at times. Please note that all the contact locations must have the same mode. You can not have one contact as proxy and another as redirect.

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Edit Third-party registration

Third party registration allows others to register for you. You must specify a valid userid of the form user@domain.

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Voice mail

Voice mail is an optional component. Ask your administrator to license sipum to avail the voicemail facility.

After clicking on the voice mail icon you can view your voice messages. The interface is similar to other web based email systems like hotmail. An example inbox is shown below.

The red pointer indicates that the message is a new message. The date is column lists the time/date when the message was received. The senders address is in "From" column. Subject is the SIP subject of the original call. If the SIP call has priority then it is printed before the subject. If there is no subject then the message number is displayed instead. The size of the message is given in both seconds and kilobytes.

User can delete the messages by selecting them; clicking on the check box in the left most column and then clicking on the "Delete Selected" link. Mails can be moved to other folders by selecting the mails and selecting the appropriate folder. The selected mails can be forwarded to a specified email address as email attachments.

Voice messages are stored as .au files or G.711 Mu Law 8kHz 8bit audio file. To play the voice message you can just click on the subject field of the appropriate message. Make sure that your browser is configured to play the MIME-type of audio/basic (or .au extension) properly. For example on Solaris with netscape, you can edit the preferences as Edit->Preferences->Navigator->Applications->New.

Description: AU 
MIMEType: audio/basic
Suffixes:.au 
Application: audioplay -i -V -v 100 -p speaker %s.

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Conferencing

Conferencing is an optional component. Ask your administrator to license sipconf to avail the conferencing facility.

Click on the conferencing icon to display the conferencing page. An example conferencing page is shown below.

The conference is identified using a conference URL. For example, if the conference URL is "test" then the users can join the conference by dialing "sip:test@domain" where domain is the domain your system is in. Restricted conferences can restrict the participation of the users in terms of who can listen or speak.

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List of users

Administrators can view the list of users by clicking on the list icon . The list gives a comprehensive listing of the registered users.

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Adding a new user

Administrator can add a new user by clicking on the new user icon . This is typically needed for adding special user ids like "tel:+12129397130", for mapping a telephone number to a user id.

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Gateway class

The gateway class menu allows an administrator to update the gateway class needed for making a call to a telephone number through a SIP-PSTN gateway. Different gateway classes can have different dialing privileges, e.g, student may be disallowed from dialing a long distance number where as faculty and staff may be allowed to do so. Every user is given a gateway class, similar to the unix group id. When a user wants to make a call to a telephone number the gateway class of the user is compared against the allowed gateway classes for that type of number. Please see the example sipd/gateways.sample and tools/canonicalize/dialplan.sample files and associated documentation.

Accounting, Rating and Billing

Accounting, rating and billing are three stages that are based on the SQL request log for the SIP server. This feature is designed to support simple re-billing, not serve as a multi-user billing interface. For example, the mechanism does not support fixed charges, multiple currencies, sending invoices or collecting credit card information. It should be sufficient for many departmental rebilling applications.

The request log function lists all requests, if invoked by the administrator, or the requests of the user logged in. The request list can be sorted according to time, request type, source and destination by clicking on the column heading. At the bottom of the page, a menu allows to limit the list to a particular time range. This feature will only work if the server has been configured to log requests to the SQL database.

Tariffs can be created and edited . A tariff specifies what a call costs per time unit. A tariff applies to a certain date range and during certain times of the day. Also, it can be restricted to a user class. The current implementation does not support tariffs that apply only on certain days, e.g., weekends and holidays, as most telecom tariffs are now time-insensitive. The prefix indicates the user name this tariff applies to, generally a phone number. This is a glob pattern, with ? representing any single character and * any number of characters. Thus, +1212* represents phone numbers in Manhattan. Note: in almost all cases, a trailing * is required. Tariff durations are automatically rounded, i.e., an end time of 23:59 is rounded to 23:59:59. There is no need to delete tariffs; if a particular tariff is no longer valid, mark its end date accordingly.

For each billing period, calls can be rated , i.e., assigned a cost. Each call is only rated once, since the billing period and cost is recorded in the call detail record. It is possible that some calls do not match any tariff and thus cannot be rated. Calls that could not be rated are listed in orange.

Finally, users can see their bills for each billing period. All calls are tallied by time, duration and destination, with a total duration and expense given.

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Config files

The administrator user can edit the configuration options from the web by clicking on the configuration icon . The configuration page allows you to edit various application specific configuration files, e.g., "sipd.conf"; as well as the global system configuration file "cinema.conf". Please refer to sipd documentation for editing "sipd.conf" file. Please refer to installation manual for editing "cinema.conf" file.

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Logging out

Users are advised to log out after they are done. If the users do not log out then the userid and password are stored in the system and next time the user visits the web page he does not have to enter the password again.

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Deleting a user

Users can themself sign-out from the system (see Edit User) or can be kicked-out by the administrator (see user list).

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