SipChannel Methods

Accepts an incoming call by sending a Session Progress Message.

This method is used to pickup an inbound call received on a Channel Resource.

Changes the RTP Type.

Changes the Stream Duplex of the audio stream. Use StreamDuplex.SendOnly to place a call on hold. Use StreamDuplex.Both to take off hold.

Reinvites the audio stream to a new Host and Port specified.

Reinvite the audio stream to a new Host, Port with a new RTP Type.

Reinvite the audio stream to a new Host, Port with a new RTP Type and Stream Duplex.

Reinvite the audio stream to a new Host, Port with a new RTP Type, Stream Duplex and RFC2833 DTMF switch.

Reinvite the audio stream to a new Host, Port with a new RTP Type, Stream Duplex, RFC2833 DTMF switch and packet time.

Dials the phone number or destination specified in the phonenumber parameter.

Dials out using the specified DestinationGroup

This method forces the Channel Resource to disconnect any current connections or calls. You may also think of this as "hanging up."

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.

This method forces a dispose of the Channel Resource object. Always do this in hang up handling to ensure clean up.

Retreives a compatible Fax Resource for this Channel. It also sets the Fax Resource Property of the Channel.

Returns the status of any current call. True if a call is connected, false if not connected.

Redirects an incoming call to an alternate address specified with the format: <sip:500@162.18.13.4>.

Redirects an incoming call to all alternate addresses specified in an array with the format: <sip:500@162.18.13.4>.

Performs an attended transfer of the current call to the target Sip Channel.

Performs an unattended transfer of the current call to the target IP Address / Phone Number.

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.

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.

Reinvite audio back to the server from another endpoint.

Issue a refresh reinvite to ensure callers are still talking.

This overload is for Voice Elements Internal Use Only! To correctly route, use the Full Route Method.

Completes a full route between two Routable Resources. Both channels then listen to each other.

This overload is for Voice Elements Internal Use Only! To correctly route, use the Half Route Method.

Completes a half route between two Routable Resources. The channel calling this method listens to the one in the parameter, the second cannot hear.

Send a NOTIFY request

Send a SUBSCRIBE request

Sets the levels for the AGC in conferencing. If all values are set to negatives, then the system wide AGC applies.

Sets the Echo Tail Delay in conferencing.

Sets the front end gain in conferencing. If set to 0, front end gain is turned off.

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.

Sets the Monitor Status element for this channel when viewed through the Voice Elements Dashboard.

Forces all Routable Resources currently listening to this resource to stop.

Instructs the channel to stop a currently in progress dial.

Forces a apecific Routable Resource currently listening to this resource to stop.

Forces this resource to stop listening to all Routable Resources.

Transfers the Channel Resource to a different application on the same Telephony Server.