SipChannel Class


SipChannel Class

Represents a SIP based channel on the Telephony Server and supports SIP speicifc fundtions like Refer, Reinvite. Inherits from Channel Resource.

Inheritance Hierarchy

SystemObject
  ClientObject
    VoiceElements.ClientRoutableResource
      VoiceElements.ClientChannelResource
        VoiceElements.ClientSipChannel

Namespace: VoiceElements.Client
Assembly: VoiceElementsClient (in VoiceElementsClient.dll) Version: 8.6.1.1

Syntax

C++

Copy

public class SipChannel : ChannelResource

Public Class SipChannel
	Inherits ChannelResource

public ref class SipChannel : public ChannelResource

type SipChannel =  
    class
        inherit ChannelResource
    end

The SipChannel type exposes the following members.

Properties

	Name	Description
	Ani	This is the originating number of the last inbound call (if available). (Inherited from ChannelResource.)
	CallConnected	The time the call connected (Inherited from ChannelResource.)
	CallEnd	The time the call was terminated. (Inherited from ChannelResource.)
	CallerIdName	This is the originating name field of the last inbound call (if available). (Inherited from ChannelResource.)
	CallNumber	The call number on the server of the current or last call. (Inherited from ChannelResource.)
	CallProgress	Gets or Sets the Call Progress Mode for this channel. (Inherited from ChannelResource.)
	CallProgressOverrides	A serializable object that provides override information to the call progress template. This is used to override Dialogic call progress settings for the next call. (Inherited from ChannelResource.)
	CallProgressTemplate	Gets or sets the call progress template to use as a basis for the next call. This is used to override Dialogic call progress settings. (Inherited from ChannelResource.)
	CallStart	The time the call was started. (Inherited from ChannelResource.)
	Conference	Represents the Conference of which this resource is a member. (Inherited from RoutableResource.)
	ConferenceAttributes	Represents the Conference Attributes of a Conference to which this resource is a member. (Inherited from RoutableResource.)
	DeviceName	The unique name of this resource / device on the server. This is useful for following, logging a call or thread. (Inherited from RoutableResource.)
	Dialog	Gets the current dialog for the channel. This is read-only.
	DialResult	Gets the Dial Result of the last call based on the Call Progress Mode setting. (Inherited from ChannelResource.)
	Dnis	This is the DNIS or the inbound number dialed of the last inbound call. (Inherited from ChannelResource.)
	DropError	The Drop Error Code of a disconnected call. (Inherited from ChannelResource.)
	DropTime	A time at which the server should automatically drop the call whether it is disonnected or not. This value is Universal Time. (Inherited from ChannelResource.)
	FaxResource	Gets the fax resource associated with this channel. To get a fax resource, call Get Fax Resource Method on this channel. (Inherited from ChannelResource.)
	GcCause	GC Cause code for a disconnected call if available. (Inherited from ChannelResource.)
	GcCauseMessage	GC Cause message for a disconnected call if available. (Inherited from ChannelResource.)
	GeneralCause	General Cause code for a disconnected call if available. (Inherited from ChannelResource.)
	GeneralCauseMessage	General Cause message for a disconnected call if available. (Inherited from ChannelResource.)
	IncomingSipHeaders	Gets the incoming SIP headers for the current call. This is read-only.
	IncomingSipRequestLine
	Listeners	A list of Routable Resources which are currently listening to this resource. (Inherited from RoutableResource.)
	ListentingTo	A Routable Resources to which this resource is currently listening. (Inherited from RoutableResource.)
	LocalCallControlAddress
	MaximumTime	Sets the maximum time in seconds to wait for a Dial to complete. Default is 30. (Inherited from ChannelResource.)
	OriginatingCallerIdName	Sets the outbound caller ID name field value.
	OriginatingPhoneNumber	This is the CallerID number sent with an oubound call when a Dial is executed. (Inherited from ChannelResource.)
	OutgoingSipHeaders	Sets the SIP headers for an outbound call.
	OverrideDestination
	OverrideRequestLine
	PortIndexer	The one based index of this channel on the server. (Inherited from ChannelResource.)
	RemoteCallControlAddress	The call control address on the remote side that is handling this call. The string format is: IP Address:Port.
	RtpAddress	Returns the address of the RTP currently in use on this channel.
	RtpCodec	Returns the name of the RTP codec currently in use on this channel.
	RtpEncryptionMode
	RtpEncryptionState
	RtpPort	Returns the port being used for RTP on this channel.
	RtpSdp	Returns the SDP (Session Description Protocol)of the RTP currently in use on this channel.
	TransferredCall	A flag to indicate that this call was transferred from another application. If true, TransferredData may contain data passed from the other application. (Inherited from ChannelResource.)
	TransferredData	Gets the data sent from another application which has invoked the Transfer Application Method to this application. (Inherited from ChannelResource.)
	TransportProtocol
	VoiceResource	The Voice Resource associated with this channel. Use this for audio functions like play, record and getting digits. (Inherited from ChannelResource.)

Top

Methods

	Name	Description
	AcceptCall	Accepts an incoming call by sending a Session Progress Message.
	Answer	This method is used to pickup an inbound call received on a Channel Resource. (Inherited from ChannelResource.)
	ChangeAudio(RtpType)	Changes the RTP Type.
	ChangeAudio(StreamDuplex)	Changes the Stream Duplex of the audio stream. Use StreamDuplex.SendOnly to place a call on hold. Use StreamDuplex.Both to take off hold.
	ChangeAudio(String, Int32)	Reinvites the audio stream to a new Host and Port specified.
	ChangeAudio(String, Int32, RtpType)	Reinvite the audio stream to a new Host, Port with a new RTP Type.
	ChangeAudio(String, Int32, RtpType, StreamDuplex)	Reinvite the audio stream to a new Host, Port with a new RTP Type and Stream Duplex.
	ChangeAudio(String, Int32, RtpType, StreamDuplex, Boolean)	Reinvite the audio stream to a new Host, Port with a new RTP Type, Stream Duplex and RFC2833 DTMF switch.
	ChangeAudio(String, Int32, RtpType, StreamDuplex, Boolean, Int32)	Reinvite the audio stream to a new Host, Port with a new RTP Type, Stream Duplex, RFC2833 DTMF switch and packet time.
	Dial(String)	Dials the phone number or destination specified in the phonenumber parameter. (Inherited from ChannelResource.)
	Dial(String, DestinationGroup)	Dials out using the specified DestinationGroup
	Disconnect	This method forces the Channel Resource to disconnect any current connections or calls. You may also think of this as "hanging up." (Inherited from ChannelResource.)
	Disconnect(Int32)	This method forces the Channel Resource to disconnect any current connections or calls. You may also think of this as "hanging up." This overload of the method requires you to specify a numeric cause code to the carrier telling the reason for the disconnect. (Inherited from ChannelResource.)
	Dispose	This method forces a dispose of the Channel Resource object. Always do this in hang up handling to ensure clean up. (Inherited from ChannelResource.)
	GetFaxResource	Retreives a compatible Fax Resource for this Channel. It also sets the Fax Resource Property of the Channel. (Inherited from ChannelResource.)
	IsConnected	Returns the status of any current call. True if a call is connected, false if not connected. (Inherited from ChannelResource.)
	Redirect(String)	Redirects an incoming call to an alternate address specified with the format: <sip:500@162.18.13.4>.
	Redirect(String)	Redirects an incoming call to all alternate addresses specified in an array with the format: <sip:500@162.18.13.4>.
	Refer(SipChannel)	Performs an attended transfer of the current call to the target Sip Channel.
	Refer(String, String)	Performs an unattended transfer of the current call to the target IP Address / Phone Number.
	Reinvite(SipChannel)	Reinvite this Sip Channel and specified Sip Channel to send RTP streams directly to each other. Note that both streams must have the same codec.
	Reinvite(String, String, UInt32, String)	Issues a reinvite for the current call. This allows you to redirect the RTP traffic to a different address and is used to reduce latency in the audio stream.
	ReinviteHome	Reinvite audio back to the server from another endpoint.
	ReinviteRefresh	Issue a refresh reinvite to ensure callers are still talking.
	RouteFull(TelephonyLinkInfo)	This overload is for Voice Elements Internal Use Only! To correctly route, use the Full Route Method. (Inherited from RoutableResource.)
	RouteFull(RoutableResource)	Completes a full route between two Routable Resources. Both channels then listen to each other. (Inherited from RoutableResource.)
	RouteHalf(TelephonyLinkInfo)	This overload is for Voice Elements Internal Use Only! To correctly route, use the Half Route Method. (Inherited from RoutableResource.)
	RouteHalf(RoutableResource)	Completes a half route between two Routable Resources. The channel calling this method listens to the one in the parameter, the second cannot hear. (Inherited from RoutableResource.)
	SendSipNotifyRequest	Send a NOTIFY request
	SendSipSubscribeRequest	Send a SUBSCRIBE request
	SetConferenceAGCLevels	Sets the levels for the AGC in conferencing. If all values are set to negatives, then the system wide AGC applies. (Inherited from RoutableResource.)
	SetConferenceECTailDelay	Sets the Echo Tail Delay in conferencing. (Inherited from RoutableResource.)
	SetConferenceFEGLevel	Sets the front end gain in conferencing. If set to 0, front end gain is turned off. (Inherited from RoutableResource.)
	SetMonitorData	Sets the Monitor Data element for this channel when viewed through the Voice Elements Dashboard. This is a collection of data elements you define to communicate from your application through the VE Dashboard. (Inherited from ChannelResource.)
	SetMonitorStatus	Sets the Monitor Status element for this channel when viewed through the Voice Elements Dashboard. (Inherited from ChannelResource.)
	StopAllListeners	Forces all Routable Resources currently listening to this resource to stop. (Inherited from RoutableResource.)
	StopDial	Instructs the channel to stop a currently in progress dial. (Inherited from ChannelResource.)
	StopListener	Forces a apecific Routable Resource currently listening to this resource to stop. (Inherited from RoutableResource.)
	StopListening	Forces this resource to stop listening to all Routable Resources. (Inherited from RoutableResource.)
	TransferApplication	Transfers the Channel Resource to a different application on the same Telephony Server. (Inherited from ChannelResource.)

Top

Events

	Name	Description
	CallProgressEvent	Triggered when a Call Progress Event is received
	DialComplete	Event Fired at the completion of a dial operation. (Inherited from ChannelResource.)
	Disconnected	Fires when a call disconnects or hangs up. Do your cleanup in this event code. (Inherited from ChannelResource.)
	NewCall	Fires when there is a new inbound call sent from the Voice Elements Server. (Inherited from ChannelResource.)
	QosEvent	Triggered when QOS(Quiality of Service) Events occur on an ongoing Sip call.
	StreamingStateEvent	Triggered when a Streaming State Event is received

Top

Fields

	Name	Description
	m_DialResult	(Inherited from ChannelResource.)

Top

Remarks

Documentation In Development

We are constantly adding features and new capabilities to The Voice Elements Platform. Therefore, our documentation of individual methods, properties and other elements is constantly being developed and improved. If you would like to know more about this element, please help us perfect our documentation and help you develop your application by doing one or more of the following:

Thank you for your continued support of Voice Elements.

Reference

VoiceElements.Client Namespace

Other Resources

Email Inventive Labs to Request Elaboration on this Feature