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asterisk_setup50

Settings.

The “Settings” module is located in the “IP telephony” menu. This module is intended for configuring the telephony server.

Port settings.

The first port settings are SIP (via UDP) and IAX, as well as the port range for incoming RTP connections. The following values are used by default:

SIP  5060
IAX  4569
RTP  10000-20000

If you set the “Automatically create a permitting rule” flag, it will add an enabling rule for the specified ports to the firewall.

SIP over TCP.

The “SIP over TCP” flag enables support for sending and receiving SIP packages over TCP on the selected port. This option shall be enabled if you use external or internal numbers configured to work via the “No Encryption (TCP)” connection type.

If you set the “Automatically create a permitting rule” flag, it will add an enabling rule for the specified port to the firewall.

Encryption.

The “Certificate for encryption (TLS and SRTP)” field allows you to select, change, or delete an encryption certificate for telephony. By default, an automatically generated certificate is selected. When the certificate is selected, the mandatory “Port for SIP over TLS” field is activated (by default, this is port 5061). If you set the “Automatically create a permitting rule” flag, it will add an enabling rule for the specified port to the firewall.

Note 1: to use encryption, you need to select the “Use encryption (TLS and SDES sRTP)” connection type in the settings of a specific phone number and configure the device on the subscriber's side.

Note 2: After changing the following settings - the SIP port via UDP/TCP/TLS - the telephony server will be restarted, and current calls will be interrupted.

General settings for the telephony server.

The SIP Channel Driver allows you to select the SIP protocol implementation module to be used by the ICS CUBE telephony server. Two channel drivers are available: chan_sip and chan_pjsip.

If you select the chan_pjsip driver, it will preserve the current behavior of the telephony module and provide almost the same functionality of the ICS CUBE telephony as if it was running on chan_sip. However, some parameters still differ. Depending on the selected channel driver, some options in the entire IP-telephony module become unavailable.

Differences in setting up external channels are as follows:

  • The insecure, canreinvite, and registration line options are only available for the chan_sip module.
  • The direct_media option is only available for the chan_pjsip module. This option determines whether media data can be transmitted directly between endpoints. If no, all RTP streams go through Asterisk.

Differences in general telephony settings:

  • The SRV lookup option is only available for the chan_sip module.

In addition, conferences with SFU and xphone video distribution mode only work on the pjsip channel driver.

Important! After you change the SIP channel driver setting, telephony server will restart and current calls will be interrupted/not processed. The current trunk registrations will not be interrupted. All internal/external trunks will need to reconnect and re-register on the server (i.e. all internal phones will need to reconnect to the ICS CUBE, and in case of problems with the telephony provider, you will need to re-connect it).

The “Timeout” field allows you to set a time period after which the telephony server will consider the caller as not answering the call and move the caller to the next set of rules. The default time is 30 seconds. Some telephony rules allow you to override this time period for a specific rule.

The “Use BLF” flag enables support for the Busy Lamp Field function, which allows real-time monitoring of the status of ATS subscribers (busy/free). Important: the end equipment (telephone) must support this function.

The “SRV lookup” flag activates DNS search for SRV records. It is not available when you select chan_pjsip as the SIP channel driver. DNS SRV records are a tool for setting logical recognizable addresses. This tool allows you to redirect calls to different points without having to change the logical address. The use of SRV records gives you access to many advantages of DNS. If you disable this option, it will be impossible to place SIP-calls based on domain names. Currently, support for SRV records in telephony is somewhat inefficient. Please note that if multiple SRV records are returned, the service will only use the first one.

Voicemail.

Call forwarding.

Unconditional call forwarding.

This option allows you to forward an incoming call intended for one subscriber to another subscriber while the call is being made. To do this, dial the unconditional forwarding number, and then the number of the other subscriber. You don't need to wait for a response from the second subscriber.

Normal forwarding.

This option allows you to forward an incoming call intended for one subscriber to another subscriber while the call is being made. To do this, dial the number for forwarding, and then the number of the other subscriber, wait for the answer of the second subscriber, and then hang up.

Call interception.

Interception within the group.

This option allows you to answer an incoming call intended for one subscriber to another subscriber, while the call is being made and the receiver is not lifted. This is convenient if the second subscriber sees that the first one is not in. To intercept a call intended for another subscriber, you need to enter *8 during the call. You can change the key combination to intercept a call in the “Pickup number” field.

Important: calls can be intercepted only within one group of phone numbers!

For example, with the following number structure:

A call received at number 155 can only be intercepted by 156 and 157. A call received at number 223 can only be intercepted by 322, 007 and 111. Number 777, located on the upper level, can intercept all numbers.

Directed interception.

This type of call interception is designed to intercept an incoming call to a specific internal number, regardless of the group of internal numbers.

To intercept a call intended for another subscriber, enter the combination * * (two asterisks - by default) + the internal number of the called subscriber from your phone during the call. You can change the key combination for directed interception in the “Number for directed interseption” field.

Faxes.

The “T38 Support” flag enables support for the T.38 standard for sending fax messages.

The “Error detection” field allows you to select the type of incoming message correction:

  • The “Redundancy” - Redundancy error correction value - correction of redundancy errors
  • The “FEC” - Forward error correction value - forward error correction
  • The “Not Specified” - do not check messages for errors.

The “Maximum datagram size” field allows you to define the maximum message size.

The “Convert received faxes to PDF” flag is used to determine the file format. By default, all fax messages will be in tiff format.

NAT.

This block is responsible for configuring the behavior of the telephony module, if it is located behind NAT.

The “Use NAT” flag enables the conversion of addresses within telephony packages. You need to specify the internal local area networks and external IP for the correct operation of this block.

You need to specify an external IP-address used for converting IP-addresses in SIP processing in the “External IP-address” field if the destination of SIP messages is outside the IP network defined in the “Local networks” field. Thus, all networks specified in this field will be considered local by the telephony server, and the rules for converting IP-addresses inside IP telephony packages will not be applied to them.

Mail server.

This block is responsible for configuring fax forwarding, notifications, and voicemail messages.

Choose which server will be used to send emails: the ICS CUBE mail server or an external SMTP server. In the first case, you just need to specify one of the email addresses that were created in the ICS CUBE in the “Sender mail” field. In the second one, you need to fill in the following fields: “SMTP server”, “Port”, “Login”, “Password” and “Sender mail”; and, if necessary, set the “SSL” flag.

Codecs.

In this section, you can select the default codecs used by the telephony module for all numbers.

The “Use” column on the left lists the codecs that are used by all numbers, unless otherwise specified in the settings of individual numbers. The order of codecs in this column matters - codecs located at the top have higher priority than those at the bottom. In other words, a list of codecs will be presented to the remote party during the establishment of a communication session in the order they appear in this list.

Codecs available but not used by the telephony module are listed in the right column - “Available codecs”.

The telephony module in the ICS CUBE supports the following codecs:

Audio:

  • G.711 a-law
  • G.711 u-law
  • G.723
  • G.726
  • G.729
  • Opus
  • GSM

Video:

  • H.264
  • VP8
  • VP9

For video calls to work, you need to add at least one video codec to the “Use” column. For xphone, it is recommended to use the following set of codecs: G.711 a-law, Opus, VP8, VP9.

Melody on hold.

This block specifies the ringtone that will be played to the caller when the call is held.

To download a ringtone, click the “Load new melody” button and select an audio file. After downloading, you can listen to the ringtone or delete it using the corresponding function buttons.

asterisk_setup50.txt · Last modified: 2020/06/19 13:39 by zog