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SipChannel Class |
Namespace: VoiceElements.Client
The SipChannel type exposes the following members.
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.) |
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>.
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![]() | Redirect(String) |
Redirects an incoming call to all alternate addresses specified in an array with the format: <sip:500@162.18.13.4>.
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![]() | 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.
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![]() | 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.
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![]() | 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.
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![]() | ReinviteHome |
Reinvite audio back to the server from another endpoint.
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![]() | ReinviteRefresh |
Issue a refresh reinvite to ensure callers are still talking.
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![]() | 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
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![]() | SendSipSubscribeRequest |
Send a SUBSCRIBE request
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![]() | 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.) |
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
|
Name | Description | |
---|---|---|
![]() | m_DialResult | (Inherited from ChannelResource.) |