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Configuring telephony connections

To configure a telephony channel, you should have a SIP connection from a third-party provider. With the help of a SIP connection, you can call the clients using your own numbers or those from your telephony provider.

SIP (Session Initiation Protocol) is a communication protocol for telephone calls. You can use the configured SIP connection not only in JAICP, but also in other services.

After you set your SIP connection from a third-party provider, you can create the connection in JAICP. Consider that only the call campaign administrator (the SYSTEM_ADMIN role) can configure telephony.

tip
More information about access and roles

Create a connection​

  1. Sign in to JAICP.

  2. On the top navigation bar, click and select SIP server connection > Add new connection.

  3. Specify SIP connection parameters in the Main settings section. By default, the 5060 port is used.

  4. If necessary, specify parameters in the Advanced settings section:

    SettingDescription
    Registration is requiredEnable the option if the registration is necessary. You may need it for receiving inbound calls from the provider if the SIP trunk requires authentication.
    Port and IP mappingEnable the option if you need to handle inbound calls to multiple DNIDs (dialed number identifiers) in one connection.
    If you do not want the source port to be checked, enable the Ignore port option.
    SIP outbound proxySpecify the host or IP address to receive call requests. The request headers will contain the host or IP address from the Main settings section.
    Use local server domain as SIP FROM headerEnable the option if you want to use the local server domain instead of the SIP URI in the FROM header.
    Check the connectionEnable the option if connection check is needed and specify a checking period in seconds (the default value is 3,600 seconds).
    Once within the specified period, JAICP will send the qualify(OPTIONS) messages to your SIP trunk. With the help of them, the SIP trunk can check whether the JAICP SIP server is available.
    ProtocolSelect the protocol: UDP or TCP.
    Sound codecsSelect the codecs. They will be used according to the order in the field.
    Number prefixSpecify the prefix to be added to the number dialed for an outbound call.
    Replace/delete the first digit in the number for outbound callsEnable the option so that the first digit in the outbound call number will be changed or deleted. If you want the digit to be changed, specify a new one.
    Calls per secondSpecify the maximum number of calls per second for outbound call campaigns.
    You may need to limit the number of calls (invite requests) to prevent SIP trunk overload. Consult with your SIP service provider for an appropriate value.
    Use direct media when transferring the callEnable the option so that the media traffic will be routed directly between the customers (via the re-invite procedure).
    SIP headersSpecify the headers. You will be able to use them in the bot script further. You can add several SIP headers.
    For more information on using SIP headers when transferring to a phone channel agent, see Reply types.

  5. Click Add a connection. You will see the connection in the list.

To add a new connection, follow the instruction again. You can create any number of connections.

caution
In the account, you can connect the same SIP trunk only once.

Edit connection data​

To edit connection data, click and select Edit. Make the changes and click Update.

Delete a connection​

To delete the connection, click and select Delete.

caution
You cannot delete a connection if there are channels that use it.