An object used to initiate an H.323, SIP, or ISDN dial-out. Once processed, there will be a corresponding plcm-participant object.
| Name | Type | Description | Attributes |
|---|---|---|---|
| name | xs:string | The display name of the participant. This value will be reflected on the MCU. | |
| dtmf-suffix | DtmfSuffix | DTMF to be outpulsed to receiving endpoint upon successful connection. | |
| audio-only | xs:boolean | True if dial out is to be audio-only. | |
| resource-priority-namespace | ResourcePriorityNamespace | Namespace parameter for the SIP Resource Priority header (namespace.value) for SIP dial outs for this participant. This is an optional value. | |
| resource-priority-value | ResourcePriorityValue | Value parameter for the SIP Resource Priority header (namespace.value) for SIP dial outs for this participant. This is an optional value. | |
| passback | Passback | User-defined value that is opaque to the system. This value will be reflected in the associated plcm-participant passback field, and allows a client to correlate the dial-out to the subsequent paritcipant. | |
| passthru | Passthru | User defined value that is opaque to the system. Note: this value can be retreived from the running call in the passthru field or after the call is terminated in the userDataC field of the call CDR. | |
| forward-dtmf-source | ForwardDtmfSource | The presence of this attribute indicates that DTMF should be forwarded from a chairperson to this participant. This attribute is set to the SIP URI of the chairperson if the URI is available, otherwise it is set to the literal string "chairperson". This attribute can be set only when a new dial-out participant is created using the REST-API request createParticipant, changing it during a call has no effect. | |
| encrypted-media | xs:boolean | Indicates whether the media stream between the conference and this participant is encrypted. | |
| auto-disconnect | xs:boolean | True if this participant is a non-live far end device (like an RMX or recording device), and can be disconnected if the only participants left in a conference are of these types. | |
| call-bit-rate | CallBitRateType | The participant call rate in kbps. The rate for h.323 and SIP can be according the the call-bit-rate-type, for ISDN it can be one of the following: automatic, 64, 128, 192, 256, 384, 512, 768, 1152, 1472, 1536, 1920. In case a different line rate is specified the DMA will round it down to one of the above values before sending it to the RMX | |
| dial-string | xs:string | The string to dial out to. The dial string can start with a scheme (e.g., "h323:", "sip:", "sips:", "isdn:"), a string with no scheme is interpreted as h323. Either dial-string or plcm-itu-phone-number is required, not both. Only dial-string is supported in this version, plcm-itu-phone-number is not used | Mandatory |
| plcm-itu-phone-number | plcm-itu-phone-number | See Definition of plcm-itu-phone-number | Mandatory |
| Name | Type | Restrictions |
|---|---|---|
| CallBitRateType | ||
| ForwardDtmfSource | xs:string | Length of value must be >=1 Length of value must be <=128 |
| Passthru | xs:string | Length of value must be >=1 Length of value must be <=512 |
| Passback | xs:string | Length of value must be >=1 Length of value must be <=512 |
| ResourcePriorityValue | xs:string | Length of value must be >=1 Length of value must be <=64 |
| ResourcePriorityNamespace | xs:string | Length of value must be >=1 Length of value must be <=64 |
| DtmfSuffix | T0 | Length of value must be >=1 Length of value must be <=64 |
| T0 | xs:string | Pattern of value must match the regular expression [\dpP0-9#\*]+ |