trunk

Syntax

Hierarchy Level

Release Information

Statement introduced in Release 10.0 of Junos OS.

Description

Configure a telephony service provider trunk to be used to route calls:

• for the SRX Series survivable call server (SRX Series SCS), when it is in control.
• for the SRX Series media gateway (SRX Series MGW) to be used by the peer call server, when it is in control.

Supported trunk types are FXO, FXS, SIP, and T1. The trunk type determines the parameters you configure. For example, for FXS and FXO trunks, you need to specify only the trunk name and its TDM interface.

You configure trunks, then add them to trunk groups, which you assign to the route patterns of a dial plan. You can use the same trunks for the SRX Series SCS and the SRX Series MGW, so it is best to give them names that are descriptive of their interfaces. For example, you might call the trunk by its interface number. You might call an FXO trunk for interface fxo-1/0/3 fxo-1/0/3.

If one trunk is busy, the next one in the trunk group is tried, and so through the configured trunks.

Options

trunk-name—Name used to refer to the telephony server provider trunk.

• direct-inward-dialing direct–inward-dialing-number—A list of DID numbers allocated for the trunk to be assigned to stations.

• trunk-type—Type of trunk to be configured. One of the following:

  • fxo

    interface–name—Name for the onboard telephony foreign exchange office (FXO) interface.

  • fxs

    interface–name—Name for the onboard telephony foreign exchange station (FXS) POTS interface.

  • sip

    sip-trunk-name—Name of the peer proxy server service provider SIP trunk.

    peer-proxy-server

    • address—
      • fqdn—Fully qualified domain name (FQDN) for the peer proxy server.
      • ipv4-address—IPv4 address for the peer proxy server.
    • auth-id authentication-id—Authentication ID used to authenticate to the peer proxy server, if required. Provided by the SIP trunk service provider.
    • auth-password authentication-password—Authentication password used to authenticate to the peer proxy server, if required. Provided by the SIP trunk service provider.
    • codec—One or more codecs, specified in order of precedence. Supported codecs are:
      • G711-A—13-bit PCM 8-kHz sample, 64 Kbps bitstream.
      • G711—14-bit PCM 8-kHz sample, 64 Kbps bitstream.
      • G729AB—CS-ACELP, 8 Kbps bitstream.
    • description description—Meaningful information about the trunk.
    • dtmf-method—Dual tone multi-frequency (DTMF) relay method to be used. One of the following:
      • inband
      • rfc-2833
      • sip-info
    • port port-number—Port number for the SIP protocol.
    • transport—Transport to be used for the SIP session. One of the following:
      • tcp—Transport Control Protocol. Because TCP can support messages of arbitrary size, it supports larger messages and provides reliability and congestion control.
      • tls—Transport Layer Security, which provides for security and data integrity of communications over the Internet Protocol (IP).
      • udp—User Datagram Protocol. A single UDP packet carries a SIP request or response. SIP does not support fragmentation, so UDP SIP messages must be smaller than the MTU size of interfaces on nodes the packet traverses.
  • t1
    • cas-group—
      • signaling—
        • fxo-loop-start or fxs-loop-start—Signaling for FXO interfaces or FXS interfaces. Loop start signaling is one of the simplest forms of CAS signaling. When the handset is picked up (the telephone goes off-hook), it closes the circuit that draws current from the telephone company (telco) and indicates a change in status, which signals the telco company to provide a dial tone. An incoming call is signaled from the telco to the handset by sending a signal in a standard on/off pattern, which causes the telephone to ring.
        • fxo-ground-start and fxs-ground-start—Signaling for FXO interfaces or FXS interfaces (fxo-ground-start or fxs-ground-start). Ground-start signaling is a technique in which a party gets a dial tone by momentarily grounding one side of a two-wire trunk.
        • em-wink-start—Ear-and-mouth-wink start is an in-band signaling method that defines a trunk-circuit side and a signaling unit side for each connection.
      • time-slots time-slots—Time slots on the T1 interface to be used for voice. Specify a single time slot or a range of time slots to be used in a specific format, such as 9 or 1–5.
    • pri-group—Identifies the T1 trunk as an ISDN PRI trunk.
      • time-slots number-ranges—Allocates the B-channel time slots defined as partition timeslots in the [edit interfaces ct1] hierarchy for use by the ISDN PRI trunk.

        Enter any combination of single slots or ranges of slots in a comma-separated list to identify the allocation for the pri-group. (For example, pri-group time-slots 1–5, 9 defines slots 1 through 5 and slot 9 as the B-channel slots in the ISDN PRI group.)

        Note: To ensure the maximum utilization of capacity in an ISDN PRI trunk, make sure that the number of B-channels specified in pri-group time-slots equals the number defined in the partition timeslots specification for the [edit interfaces ct1] hierarchy. When a pri-group is defined, CT1 time slots that are not allocated as pri-group time-slots must remain unused and cannot be allocated to another group.

    • tdm-interface tdm-interface-name—Name of the timed data multiplexing (TDM) interface for the defined CAS group or ISDN PRI group.

Usage Guidelines

For configuration instructions and examples, see the Junos OS Integrated Convergence Services Configuration and Administration Guide.

Required Privilege Level

services—To view this statement in the configuration.

services–control—To add this statement to the configuration.