4.7 FTP Server
4.8.1 Phone management
Registered Phones
In (Figure 4.41) it is possible to see the registered IPBrick VoIP clients (IP telephones, workstations + softphone). In section Machine Management you find the description of the menu to insert the VoIP machines.
It is also possible to register phones in:
Advanced Configurations - Telephony - Registered Phones
This option is valid, if it isn’t necessary to attribute a specific IP address to the phone. It is possible to add a phone just by filling the field relating the name and the access password. This assuming that DNS is working correctly.
Alternative addresses
As you can see in Figure 4.42 , to each telephone (either a hardware tele-phone or a software teletele-phone) may be associated several alternative addresses.
An alternative address is another name (or number) to reach the telephone. This
16Session Initiation Protocol
Figure 4.41: VoIP - Registered Phones
functionality is very useful when there are telephones from which you can only dial numbers.
Figure 4.42: VoIP - Alternative Addresses
Example: There is an IP telephone with the name phone01. Through the site myipbrick, an user called John Smith associates to this telephone, placing
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in the SIP URL the address phone01. An alternative address is also created for that telephone, with the name 5050. From that moment on, the user John Smith may be reached either through the phone01 or 5050, but the main idea here is to contact the user simply by [email protected].
In top menu there is a option to insert new alternative addresses. As already mentioned, these can have two types:
• Phone name: It is necessary to choose between the telephones in IPBrick, which one do you want to associate to an alternative address;
• New phone alternative address: Insert the alternative address of the telephone.
To confirm the insertion, it is necessary to click in the Insert button.
SIP URL’s
As already mentioned, it is also possible to associate a certain telephone (num-ber or name) to an internal user of the network. The association is made from the users email address in the field SIP URL. This operation is made through the site https://myipbrick.domain.com. This way, to contact a certain user all you have to do is call him/her through his/her email. The call shall be made, and the one who’s calling knows which device the addressee shall use (mobile phone, softphone, analogic/digital telephone).
4.8.2 Services
This section allows to configure all the IP PBX functionalities slitted into inbound and outbound services.
Inbound
Call Groups
In this interface (Figure 4.43) is possible to define answering groups, i.e., a group of telephones which shall ring simultaneously when the access to the group is made. To define a group it is necessary to fulfil:
• Name: Name for the group;
• Caller ID: Possibility to use a specific caller ID for this service;
• Direct access: List of numbers/addresses that will call this service. We have tree options and it’s possible to use many direct access’s;
– DID: If the IPBrick has a ISDN telephony card, the DID (Direct Inward Dial) will be the direct PSTN number that will call this service;
– ANA: If the IPBrick has a analogic telephony card, will be the direct PSTN number that will call this service;
– SIP: It’s the specific SIP address that will call this service;
• Group Members
– Internal: Internal SIP phones that belong to the group;
– External: External phones (SIP, PSTN number etc) that belong to the group.
Figure 4.43: VoIP - Call groups
Attendance seq.
In this section it is possible to define an answering sequence, or see/change/remove the already defined sequences. To add a new sequence it is necessary to click In-sert, define a name for the sequence, select if the voicemail is active or not and in Direct Access add the addresses DID/SIP/ANA of the telephones by which the sequence shall be activated.
If you intend to add a Direct Access for an extension defined in IPBrick, it is possible to choose SIP and select the extension in the address. In Sequence is possible to add the telephones which shall ring by the desired order and the time in which each one of them plays till the next one.
To define a attendance seq. it is necessary to fill (Figure 4.44):
• Name: Name for the attendance seq;
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• Caller ID: Possibility to use a specific caller ID for this service;
• Voicemail enabled: Enables the voicemail for the sequence;
• Direct access: List of numbers/addresses that will call this service. We have tree options and it’s possible to use many direct access’s;
– DID: If the IPBrick has a ISDN telephony card, the DID (Direct Inward Dial) will be the direct PSTN number that will call this service;
– ANA: If the IPBrick has a analogic telephony card, will be the direct PSTN number that will call this service;
– SIP: It’s the specific SIP address that will call this service;
• Sequence positions
– Location Internal: Internal SIP phones that belong to the sequence;
– Location External: External phones (SIP, PSTN number etc) that belong to the sequence;
– Timeout: Timeout in seconds, be default 25.
Figure 4.44: VoIP - Sequence definitions
A attendance sequences list can be viewed at Figure 4.45.
IVR Attendance
In this section is possible to define interactive answering menus (Figure 4.46).
You need to click Insert to add a new one:
Figure 4.45: VoIP - Attendance sequences list
• Name: Choose a name for IVR;
• Direct access: List of numbers/addresses that will call this service. We have tree options and it’s possible to use many direct access’s;
– DID: If the IPBrick has a ISDN telephony card, the DID (Direct Inward Dial) will be the direct PSTN number that will call this service;
– ANA: If the IPBrick has a analogic telephony card, will be the direct PSTN number that will call this service;
– SIP: It’s the specific SIP address that will call this service;
• Number of desired shortcuts: Choose how many options does the menu have;
• Shortcuts: What type of destiny to give (according to the pressed key):
– Phone: To call to a internal telephone;
– IVR: To go to an interactive answering sub-menu;
– Conference: To connect to a conference;
– Scheduler: To connect to a scheduler;
– Group: To ring the telephones of a group;
– Sequence: To activate an answering sequence;
– SIP address: To call a SIP telephone;
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– DISA: It allows someone outside the central to connect as if he/she is directly connected to the central;
– Call queue: To make the call enter a waiting line;
• Attendance message: It allows the selection of an answering message. Can be a .mp3 or .wav file;
• Number of message repetitions: Number of attendance messages replays;
• Redirect automatically when no option has been dialed: As Yes if no DTMF pressed it can redirect the call directly to:
– Phone: To call to a internal telephone;
– IVR: To go to an interactive answering sub-menu;
– Conference: To connect to a conference;
– Scheduler: To connect to a scheduler;
– Group: To ring the telephones of a group;
– Sequence: To activate an answering sequence;
– SIP address: To call a SIP telephone;
– DISA: It allows someone outside the central to connect as if he/she is directly connected to the central;
– Call queue: To make the call enter a waiting line;
Figure 4.46: VoIP - IVR attendance configuration
Call Conference
In this interface (Figure 4.47) is possible to create conferences. To create a simple static conference just click Insert:
• Name: The conference name;
• Numeric identifier: Numeric identifier for the conference. It’s only a internal identifier for the VoIP server;
• PIN: Code which shall allow the users to connect to the conference;
• Administrator PIN: Conference code for the administrator;
• Direct access: List of numbers/addresses that will call this service. We have tree options and it’s possible to use many direct access’s;
– DID: If the IPBrick has a ISDN telephony card, the DID (Direct Inward Dial) will be the direct PSTN number that will call this service;
– ANA: If the IPBrick has a analogic telephony card, will be the direct PSTN number that will call this service;
– SIP: It’s the specific SIP address that will call this service.
Figure 4.47: VoIP - Call conference insertion
It is also possible to allow the creation of dynamic conferences. For that, it is necessary to click Dynamic Conferences (Figure 4.48), modify the option Active to Yes and insert the address(es) and/or number(s) for the Direct Accesses (Figure 4.49). At dynamic conferences, when someone call to the direct access it’s possible to enter automatically a existant conference or to create a new one.
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Figure 4.48: VoIP - Call conference list
Figure 4.49: VoIP - Dynamic call conferences
Call Parking
Here (Figure 4.50) is possible to activate or deactivate the option of calls on hold.
If this option is activated, it is necessary to define an extension to place the calls on hold, as well the virtual extensions in which calls are going to be placed (Figure 4.51). To accede to these calls later is necessary to insert in the telephone keypad the ”#” plus the virtual extension were the call was parked.
Figure 4.50: VoIP - Call Parking
Figure 4.51: VoIP - Call Parking - Modify
Scheduling
This option (Figure 4.52) allows to define the behavior of the IP PBX for all the day. Usually this is the most important inbound service because from here, we are able to call all the others configured services.
It is necessary to click option Insert (Figure 4.53) and configure the first
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Figure 4.52: VoIP - Scheduling
parameters:
• Name: The name for the scheduler;
• Direct access: List of numbers/addresses that will call this service. We have tree options and it’s possible to use many direct access’s;
– DID: If the IPBrick has a ISDN telephony card, the DID (Direct Inward Dial) will be the direct PSTN number that will call this service;
– ANA: If the IPBrick has a analogic telephony card, will be the direct PSTN number that will call this service;
– SIP: It’s the specific SIP address that will call this service.
Next, it is necessary to add rules for this scheduler. For that:
• Click in the scheduler name;
• Click Insert;
• Choose the type of action to be executed;
• Choose the period to be executed.
Fields explanation:
• Destination type: Where shall the call be routed if the rule defined next is equalled. Options:
– Phone: To call to a internal telephone;
– IVR: To go to an interactive answering sub-menu;
– Conference: To connect to a conference;
– Scheduler: To connect to a scheduler;
– Group: To ring the telephones of a group;
– Sequence: To activate an answering sequence;
– SIP address: To call a SIP telephone;
– DISA: It allows someone outside the central to connect as if he/she is directly connected to the central;
– Call queue: To make the call enter a waiting line;
• Destination: Telephone address or specific service name were the call shall be routed;
• Hours: Beginning and end hour, from the timetable in which the rule shall be valid (format hh:mm at each field);
• Weekdays: Weekdays in which the rule shall be valid. If not chosed it will use all days;
• Month days: Days of the month in which rule shall be verified. If not chosed it will use all;
• Months: Months in which the rule shall be valid. If not chosed it will use all months;
Figure 4.53: VoIP - Scheduling - Insert rules
NOTE: If you don’t select any hour or days of the week/month, hour or months, the rule shall be valid respectively for all the day. A rule like this one is
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called the default rule;
At Figure 4.54 we can see an example of a scheduling implementation. You can see that the rule 4 is used from 19:01 to 08:59, because is the default time. It will call a simple IVR with a voice message telling that nobody is at the company to answer the phone.
Figure 4.54: VoIP - Scheduling - Rules list
DISA
DISA17 (Figure 4.55) is a service that allows that someone that is not directly connected to IPBrick or the PBX central, to obtain an internal call sign and execute calls as if he/she was directly connected to the internal network. The user calls the access number to DISA and he/she should type a password followed by the key ”#”. If the password is correct, the user shall hear the sign indicating that he/she may dial the number. You can also enjoy this service without a password if you want to. The fields necessary to configure a DISA are:
• Name: Name for DISA;
• Direct access: List of numbers/addresses that will call this service. We have tree options and it’s possible to use many direct access’s;
– DID: If the IPBrick has a ISDN telephony card, the DID (Direct Inward Dial) will be the direct PSTN number that will call this service;
– ANA: If the IPBrick has a analogic telephony card, will be the direct PSTN number that will call this service;
17Direct Inward System Access
– SIP: It’s the specific SIP address that will call this service.
• PIN authentication: It allows the introduction of a password to enable the dialling through DISA;
• Password: PIN password;
• Allowed caller ID’s: Callers identifiers list which may accede to this ser-vice. Insert only one by line.
Figure 4.55: VoIP - DISA - Insert
Call queues
Here (Figure 4.56) it is possible to define waiting lines. When calling to the telephone defined in Direct Access the caller shall be placed on hold if there is another call to be answered. An answering message may be defined which shall be heard when the call is on hold. It is also possible to choose messages by default in Select queue information from the line which may inform the caller about his/her position in the line and the time interval between those messages.
The settings where we hit insert are the following ones:
• Name: Name of queue;
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• Direct access: List of numbers/addresses that will call this service. We have tree options and it’s possible to use many direct access’s;
– DID: If the IPBrick has a ISDN telephony card, the DID (Direct Inward Dial) will be the direct PSTN number that will call this service;
– ANA: If the IPBrick has a analogic telephony card, will be the direct PSTN number that will call this service;
– SIP: It’s the specific SIP address that will call this service.
• Queue weight: Queue’s priority.
• Maximum number of queued calls: Maximum number defined of calls on hold. ’0’ defines an unlimited number;
• Define maximum waiting time: It is possible to define the maximum wait-ing time. For that it is necessary to click option Yes, select the maximum time in seconds and the type of routing to do if the time is exceeded as well as the final destiny;
• Phone attendance timeout: Period of time (seconds) at the end of which the caller shall be put on hold if the call is not answered, even if there is no one else on hold;
• Welcome message file: Select the message to be presented when someone enters the waiting line;
• Select queue information message: Select some of these messages to in-form about the position in the waiting line or the estimated waiting time.
Messages: ”You are now first in line”, ”There are”, ”calls waiting”, ”The current estimated holdtime is”, ”minutes”, ”seconds”, ”Thank you for your patience”, ”less than” ,”hold time” ,”All phones busy / wait for next”;
• Time interval between queue information messages: If some informa-tive message is selected, is possible to select the time (seconds) between messages;
• Attendance policy: How the waiting line answering telephones should an-swer the calls:
– Ring all: All available telephones ring until one of them answers;
– Random: One of the available telephones rings by chance;
– Round Robin: Each telephone rings at the time;
– Round Robin with memory: Each telephone rings at the time, but it remembers which was the last one to ring;
– Least recently called phone: Will ring the telephone that rung a long time ago;
– Phone with fewest completed calls: Will ring the telephone with less answered calls.
• Play message when call is answered: If Yes a message will be played before the call is answered;
Figure 4.56: VoIP - Call queue definitions
When a call queue is inserted there are the following options at the top: Back, Modify, Delete and Members. So the next step is to define what IP phones or/and LDAP users will be associated to the call queue. Clicking Members you will get a list of phones and users, like shown at Figure 4.57.
At Call queues - Agents, we have a list of IPBrick LDAP users. A user can be defined as a call queue agent. To configure one agent click at one name, choose Yes and configure:
• Login: Number used to enter dynamically a call queue;
• Waiting mode
– Music on hold: The phone will be immediately part of the call queue.
The user will listening music until a call is received;
– Callback: Only if the agent receive a call from the call queue, the phone will ring;
• With PIN?: If Yes the user must enter a PIN after the login number;
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Figure 4.57: VoIP - Call queue members
Outbound
Access Classes
It is possible to define access rules for the existing telephones. For that it is necessary to click on the connection Insert and fulfil the following fields (Figure 4.59):
• Name: The access class name;
• Unlock code: Code to deactivate temporarily a access class;
• Prefixes: It allows to add to the authorized prefixes list the prefixes which may be used in the telephones under the access rules. By default all the calls are blocked except the Authorized prefixes;
• Numbers: In Politics by default it is possible to block the traffic for any number or let it pass by default (Block/Authorize, respectively) and then, if there are some exceptions, it is possible to indicate an exception number by line. You can use wildcards at the exceptions;
• Domains: In the same way it is possible to authorize or block the access to certain numbers, it is also possible with VoIP domains at Internet.
Figure 4.58: VoIP - Call queue agents
To confirm and create a defined rule, click Insert. Now it is possible to add the members under that rule, clicking the name of the rule and then Members (Figure 4.60). To remove or add SIP phones to the access class you only have to click the buttons or respectively.
Speed Dial
The speed dial allow the association between an internal address and a tele-phone external to the organization. That is, the users call an internal number (or address) and this is associated to a telephone external to the organization. Exam-ple: An external alternative address of the telephone [email protected] is created for the destiny address [email protected]. This way, whenever you dial internally 44, the call shall be re-addressed to [email protected].
Choosing Speed Dial and clicking Insert we have two fields (Figure 4.61):
• Phone Address: Will be the external number or address to call;
• Phone Address: Will be the external number or address to call;