Update SIP Voice IN Trunk
update_sip_trunkUpdate one of the authenticated customer's SIP Voice IN Trunks (inbound voice trunks). Pass the trunk id and ONLY the fields to change — never invent values for fields the user did not ask about; untouched fields keep their current values. Fields have the same semantics as in create_sip_trunk. For PSTN trunks use update_pstn_trunk; for phone.systems™ trunks use update_phone_systems_trunk. Returns the updated trunk attributes (passwords masked) or a readable error.
Input Schema
| Name | Required | Description | Default |
|---|---|---|---|
| id | Yes | UUID of the SIP Voice IN Trunk to update. | |
| pop | No | DIDWW Point of Presence the trunk is served from. Null unpins the trunk and lets DIDWW pick. | |
| host | No | New host part of R-URI (SIP host/IP). | |
| name | No | New trunk name. | |
| port | No | New port part of R-URI. Null resolves the SRV/A record instead. | |
| codecs | No | New codec list, in order of preference. Replaces the whole list, so keep "telephone-event" in it unless the user asked otherwise — it carries RFC 2833 DTMF, and dropping it stops DTMF from reaching the customer equipment. | |
| weight | No | Load-balancing weight among trunk-group members of the same priority; higher is preferred (default 65535). | |
| priority | No | Priority of this trunk; DIDWW contacts the lowest-numbered priority first. Range 0-65535. | |
| rtp_ping | No | Enable/disable RTP keep-alive packets. | |
| username | No | New user part of R-URI ("{DID}" pattern supported). Null removes it. | |
| auth_user | No | New username for outgoing SIP authentication. | |
| cli_format | No | Format of the caller ID sent to the customer equipment (default e164). | |
| cli_prefix | No | Prefix prepended to the caller ID. Letters, digits, "+" and "#" only, up to 7 characters. Null removes the prefix. | |
| cnam_lookup | No | Enable/disable CNAM lookup on the trunk. | |
| description | No | Description of the Voice IN Trunk. Null clears it. | |
| rtp_timeout | No | New number of seconds without inbound RTP before the call is dropped. | |
| sip_timer_b | No | New INVITE transaction timeout in MILLISECONDS. | |
| sst_enabled | No | Enable/disable SIP Session Timers. Enabling requires sst_min_timer and sst_max_timer. | |
| auth_enabled | No | Enable/disable outgoing SIP digest authentication (enabling requires auth_user and auth_password). | |
| resolve_ruri | No | Resolve the R-URI host to an IP before sending the INVITE. | |
| auth_password | No | New password for outgoing SIP authentication. | |
| max_transfers | No | New maximum number of SIP transfers (REFER) to follow. | |
| sst_max_timer | No | New maximum session timer in seconds; must be >= sst_min_timer. | |
| sst_min_timer | No | New minimum session timer in seconds. | |
| auth_from_user | No | New From-header user part. Null removes it. | |
| capacity_limit | No | Maximum number of simultaneous calls for the Voice IN Trunk. Null lifts the limit. | |
| rx_dtmf_format | No | New format for DTMF received from the DIDWW network. | |
| sst_accept_501 | No | Treat a 501 answer to the session refresh as success. | |
| trunk_group_id | No | Voice IN Trunk Group UUID to assign the trunk to (or null to detach). | |
| tx_dtmf_format | No | New format for DTMF sent to the customer equipment. | |
| allowed_rtp_ips | No | New RTP source restriction (CIDR notation, e.g. "203.0.113.0/24"). Replaces the whole list; up to 10 entries, and 0.0.0.0/0 and ::/0 are rejected. Pass null to remove the restriction and accept RTP from any address. | |
| ringing_timeout | No | Seconds to wait for a 200 OK after a 18x ringing response before ending the routing attempt with the "Ringing timeout" disconnect code. Null restores the platform default. | |
| use_did_in_ruri | No | Only with enabled_sip_registration: put the called DID in the R-URI user part. | |
| auth_from_domain | No | New From-header domain part. Null removes it. | |
| stir_shaken_mode | No | New STIR/SHAKEN attestation passing mode. | |
| max_30x_redirects | No | New maximum number of 3xx redirects to follow. | |
| src_number_list_id | No | Voice IN Number List UUID to apply as the caller-ID filter (or null to detach). | |
| sst_refresh_method | No | New SIP method used to refresh the session. | |
| transport_protocol | No | New SIP transport protocol. | |
| force_symmetric_rtp | No | Send RTP back to the source address of the received stream instead of the SDP address. | |
| sst_session_expires | No | New Session-Expires value in seconds; must be between sst_min_timer and sst_max_timer. Null clears it. | |
| diversion_relay_mode | No | New Diversion header relay mode. | |
| diversion_inject_mode | No | New Diversion header inject mode. | |
| media_encryption_mode | No | New SRTP media encryption mode. | |
| dns_srv_failover_timer | No | New DNS SRV failover time in milliseconds; must not exceed sip_timer_b. | |
| enabled_sip_registration | No | Switch the trunk to/from SIP-registration mode. When enabling, host/port are cleared and registration credentials are generated server-side; when disabling, set a host. | |
| network_protocol_priority | No | New IP version preference when resolving the host. | |
| symmetric_rtp_ignore_rtcp | No | With force_symmetric_rtp, ignore RTCP when learning the source address. | |
| rerouting_disconnect_codes | No | New list of SIP response codes that trigger rerouting ("486" Busy Here, "503" Service Unavailable, ...), plus "Ringing timeout" — the no-answer disconnect reason, which has no SIP code and is not a timeout setting. Replaces the whole list; omit the argument to leave it untouched, or pass null to go back to the platform default set. |