Voice IN Trunk Object

Description

Json API object with type voice_in_trunks. You can get several type of trunks: SIP, PSTN, H323, IAX2.

Request

URI Parameters

Name	Type	Is Required?	Description
id	string	Yes	Unique ID identifier of the Trunk.

Data Attributes

Name	Type	Description
priority	integer	The priority of this target host. DIDWW will attempt to contact the target trunk with the lowest-numbered priority; target trunk with the same priority will be tried in an order defined by the weight field. The range is 0-65535. See RFC 2782 for more details.
weight	integer	A trunk selection mechanism. The weight field specifies a relative weight for entries with the same priority. Larger weights will be given a proportionately higher probability of being selected. The range of this number is 0-65535. In the presence of records containing weights greater than 0, records with weight 0 will have a very small chance of being selected. See RFC 2782 for more details.
capacity_limit	integer	Maximum number of simultaneous calls for the trunk.
ringing_timeout	integer	After which it will be end transaction with internal disconnect code Ringing timeout if the call was not connected.
name	string	Friendly name of the trunk.
cli_format	string	RAW - Do not alter CLI (default). E164 - Attempt to convert CLI to E.164 format. Local - Attempt to convert CLI to Localized format. CLI format conversion may not work correctly for phone calls originating from outside the country of that specific DID.
cli_prefix	string	You may prefix the CLI with an optional + sign followed by up to 6 characters, including digits and #.
description	string	Optional description of the trunk.
configuration	One of sip_configurations, pstn_configurations	Trunk configuration complex object.
created_at	DateTime	Trunk created at DateTime

Attributes Configuration

SIPPSTN

Name	Type	Nullable	Is Required?	Description
type	sip_configurations	No	Yes	SIP configuration complex object.
attributes	sip_configuration_attributes	No	Yes	SIP configuration attributes object.

Configuration Attributes

SIPPSTN

Name	Type	Nullable	Is Required?	Description
username	string	No	Yes	User part of R-URI in INVITE request. You also may use “{DID}” pattern which will be replaced by called DID number in E164 format. For example, you can set Username to “+{DID}”; if you wish to have it in +E164 format
host	string	No	Yes	Host part of R-URI in INVITE request.
transport_protocol_id	integer	No	No	Transport protocol ID. Possible values: 1 - TCP 2 - UDP 3 - TLS
media_encryption_mode	string	No	No	Media encryption mode. See RFC4568 about SRTP SDES, RFC5764 about SRTP DTLS, and RFC6189 about ZRTP for more details. Possible values: ‘disabled’ - Disabled ‘srtp_sdes’ - SRTP SDES ‘srtp_dtls’ - SRTP DTLS ‘zrtp’ - ZRTP
stir_shaken_mode	string	No	No	Stir/Shaken mode. See STIR/SHAKEN for more details. Possible Values: ‘disabled’ - Do not send identity ‘original’ - Transit Identity header as is ‘pai’ - Add PAI, P-Attestation-Indicator, P-Origination-ID ‘original_pai’ - Transit Identity Header as is + Add PAI, P-Attestation Indicator, P-Origination-ID ‘verstat’ - Add P-Stir-Verstat, P-Attestation-Indicator, P-Origination-ID
auth_user	string	No	No	Optional authorization user for the SIP server.
auth_password	string	No	No	Optional authorization password for the SIP server.
auth_from_user	string	No	No	Specify user in a from field instead of CallerID (overrides CallerID). Some equipment require from; to be equivalent to Auth user.
auth_from_domain	string	No	No	Sets default from domain in SIP messages. Some equipment may require specific From Domain.
sst_refresh_method_id	integer	No	No	SIP method which will be used for session update. See RFC 4028 for more details. Possible values: 1 - Invite 2 - Update 3 - Update fallback Invite
sip_timer_b	integer	No	No	INVITE transaction timeout (Default 8000ms). See RFC 3261 Section 17.1.1.2 for more details.
dns_srv_failover_timer	integer	No	No	Invite transaction timeout for each of gateways with DNS SRV rerouting (Default 2000ms).
rtp_ping	integer	No	No	Use RTP PING when connecting a call. After establishing the call, DIDWW will send empty RTP packet RTP PING. It is necessary if both parties operate in Symmetric RTP / Comedia mode and expect the other party to start sending RTP first.
rtp_timeout	boolean	No	No	Disconnect the call if the RTP packets do not arrive within the specified time.
allowed_rtp_ips	Array of strings	No	No	The allowed RTP IPs. Array from 0 to 10 items: IPv4 or IPv6, single or subnet.
sst_min_timer	integer	No	No	Minimal SIP Session timer value (Default 600 seconds). See RFC 4028 for more details.
sst_max_timer	integer	No	No	Maximal SIP Session timer value (Default 900 seconds). See RFC 4028 for more details.
sst_session_expires	integer	No	No	Session-Expires header value. Optional, should be in range with sst_min_timer and sst_max_timer. See RFC 4028 for more details.
port	integer	No	No	Port part of R-URI in INVITE request (is not mandatory). If port is null, SRV record will be resolved (or A record if SRV is unavailable).
rx_dtmf_format_id	integer	No	No	The method id for receiving DTMF signals from customers equipment. Possible values: 1 - RFC 2833 2 - SIP INFO application/dtmf-relay OR application/dtmf 3 - RFC 2833 OR SIP INFO
tx_dtmf_format_id	integer	No	No	The method of sending DTMF signals to customers equipment. Possible values: 1 - Disable sending 2 - RFC 2833 3 - SIP INFO application/dtmf-relay 4 - SIP INFO application/dtmf
force_symmetric_rtp	boolean	No	No	Forced to work in Symmetric RTP / COMEDIA mode.
symmetric_rtp_ignore_rtcp	boolean	No	No	Avoid switching RTP session based on RTCP packet while working in Symmetric RTP / COMEDIA. Only RTP packets will be considered.
sst_enabled	boolean	No	No	Enable SIP Session timers customization. SIP session timers are used to make sure that a session (dialog) is still alive, even though there may have been a long time since the last in-dialog message. If the other end is not responding, the dialog will be hung up automatically. SIP session timers need to be supported by all end points for it to work. It’s a SIP extension, standardized by the IETF. See RFC 4028 for more details.
sst_accept_501	boolean	No	No	Do not drop the call after receiving SIP 501 response for non-critical messages.
auth_enabled	boolean	No	No	Enable authorization for the SIP server.
resolve_ruri	boolean	No	No	Replace host part of the R-URI by resolved IP address.
rerouting_disconnect_code_ids	array	No	No	Rerouting disconnect codes.
codec_ids	array	No	No	Codecs
transport_protocol_id	integer	No	No	The transport layer that will be responsible for the actual transmission of SIP requests and responses (1 - UDP, 2 - TCP).
max_transfers	integer	No	No	Max count of the REFER requests.
max_30x_redirects	integer	No	No	Max count of 301/302 redirects.