|
public Hangup ( |
||
| sipTransport | ||
| outboundProxy | ||
| return | void | |
private void UACCallAnswered(ISIPClientUserAgent answeredUAC, SIPResponse answeredResponse)
{
try
{
// Remove the current call from the pending list.
lock (m_switchCalls)
{
m_switchCalls.Remove(answeredUAC);
}
if (m_switchCallTransactions != null && answeredUAC.ServerTransaction != null)
{
m_switchCallTransactions.Add(answeredUAC.ServerTransaction);
}
if (answeredResponse != null && answeredResponse.StatusCode >= 200 && answeredResponse.StatusCode <= 299)
{
#region 2xx final response.
if (!m_callAnswered && !m_commandCancelled)
{
// This is the first call we've got an answer on.
m_callAnswered = true;
m_answeredUAC = answeredUAC;
AnsweredSIPResponse = answeredResponse;
SIPDialogueTransferModesEnum uasTransferMode = SIPDialogueTransferModesEnum.Default;
if (m_answeredUAC.CallDescriptor.TransferMode == SIPDialogueTransferModesEnum.NotAllowed)
{
answeredUAC.SIPDialogue.TransferMode = SIPDialogueTransferModesEnum.NotAllowed;
uasTransferMode = SIPDialogueTransferModesEnum.NotAllowed;
}
else if (m_answeredUAC.CallDescriptor.TransferMode == SIPDialogueTransferModesEnum.BlindPlaceCall)
{
answeredUAC.SIPDialogue.TransferMode = SIPDialogueTransferModesEnum.BlindPlaceCall;
uasTransferMode = SIPDialogueTransferModesEnum.BlindPlaceCall;
}
else if (m_answeredUAC.CallDescriptor.TransferMode == SIPDialogueTransferModesEnum.PassThru)
{
answeredUAC.SIPDialogue.TransferMode = SIPDialogueTransferModesEnum.PassThru;
uasTransferMode = SIPDialogueTransferModesEnum.PassThru;
}
/*else if (m_answeredUAC.CallDescriptor.TransferMode == SIPCallTransferModesEnum.Caller)
{
answeredUAC.SIPDialogue.TransferMode = SIPDialogueTransferModesEnum.NotAllowed;
uasTransferMode = SIPDialogueTransferModesEnum.Allowed;
}
else if (m_answeredUAC.CallDescriptor.TransferMode == SIPCallTransferModesEnum.Callee)
{
answeredUAC.SIPDialogue.TransferMode = SIPDialogueTransferModesEnum.Allowed;
uasTransferMode = SIPDialogueTransferModesEnum.NotAllowed;
}
else if (m_answeredUAC.CallDescriptor.TransferMode == SIPCallTransferModesEnum.Both)
{
answeredUAC.SIPDialogue.TransferMode = SIPDialogueTransferModesEnum.Allowed;
uasTransferMode = SIPDialogueTransferModesEnum.Allowed;
}*/
if (CallAnswered != null)
{
logger.Debug("Transfer mode=" + m_answeredUAC.CallDescriptor.TransferMode + ".");
CallAnswered(answeredResponse.Status, answeredResponse.ReasonPhrase, null, null, answeredResponse.Header.ContentType, answeredResponse.Body, answeredUAC.SIPDialogue, uasTransferMode);
}
// Cancel/hangup and other calls on this leg that are still around.
CancelNotRequiredCallLegs(CallCancelCause.NormalClearing);
}
else
{
// Call already answered or cancelled, hangup (send BYE).
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Call leg " + answeredUAC.CallDescriptor.Uri + " answered but call was already answered or cancelled, hanging up.", m_username));
SIPDialogue sipDialogue = new SIPDialogue(answeredUAC.ServerTransaction, m_username, m_adminMemberId);
sipDialogue.Hangup(m_sipTransport, m_outboundProxySocket);
}
#endregion
CallLegCompleted();
}
else if (answeredUAC.SIPDialogue != null)
{
// Google Voice calls create the dialogue without using a SIP response.
if (!m_callAnswered && !m_commandCancelled)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Call leg for Google Voice call to " + answeredUAC.CallDescriptor.Uri + " answered.", m_username));
// This is the first call we've got an answer on.
m_callAnswered = true;
m_answeredUAC = answeredUAC;
if (CallAnswered != null)
{
CallAnswered(SIPResponseStatusCodesEnum.Ok, null, null, null, answeredUAC.SIPDialogue.ContentType, answeredUAC.SIPDialogue.RemoteSDP, answeredUAC.SIPDialogue, SIPDialogueTransferModesEnum.NotAllowed);
}
// Cancel/hangup and other calls on this leg that are still around.
CancelNotRequiredCallLegs(CallCancelCause.NormalClearing);
}
else
{
// Call already answered or cancelled, hangup (send BYE).
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Call leg for Google Voice call to " + answeredUAC.CallDescriptor.Uri + " answered but call was already answered or cancelled, hanging up.", m_username));
answeredUAC.SIPDialogue.Hangup(m_sipTransport, m_outboundProxySocket);
}
}
else if (answeredResponse != null && answeredResponse.StatusCode >= 300 && answeredResponse.StatusCode <= 399)
{
ProcessRedirect(answeredUAC, answeredResponse);
}
else if (answeredResponse != null)
{
// This call leg failed, record the failure status and reason.
m_lastFailureStatus = answeredResponse.Status;
m_lastFailureReason = answeredResponse.ReasonPhrase;
if (m_switchCallTransactions != null && answeredUAC.ServerTransaction != null)
{
m_switchCallTransactions.Add(answeredUAC.ServerTransaction);
}
CallLegCompleted();
}
}
catch (Exception excp)
{
logger.Error("Exception ForkCall UACCallAnswered. " + excp);
}
}
/// <summary>
/// Performs an attended transfer based on a REFER request with a Replaces parameter on the Refer-To header.
/// </summary>
/// <param name="dialogue">The dialogue matching the the REFER request headers (Call-ID, To tag and From tag).</param>
/// <param name="referTransaction">The REFER request.</param>
/// <param name="localEndPoint">The local SIP end point the REFER request was received on.</param>
/// <param name="remoteEndPoint">The remote SIP end point the REFER request was received from.</param>
private void ProcessAttendedRefer(SIPDialogue dialogue, SIPNonInviteTransaction referTransaction, SIPRequest referRequest, SIPEndPoint localEndPoint, SIPEndPoint remoteEndPoint)
{
try
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Initiating attended transfer.", dialogue.Owner));
SIPUserField referToField = SIPUserField.ParseSIPUserField(referRequest.Header.ReferTo);
if (referToField == null)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Error on transfer, could not parse Refer-To header: " + referRequest.Header.ReferTo + ".", dialogue.Owner));
SIPResponse errorResponse = SIPTransport.GetResponse(referRequest, SIPResponseStatusCodesEnum.BadRequest, "Could not parse Refer-To header");
referTransaction.SendFinalResponse(errorResponse);
}
else
{
string replaces = referToField.URI.Headers.Get(m_referReplacesParameter);
SIPDialogue replacesDialogue = GetDialogue(replaces);
if (replacesDialogue == null)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Could not locate the dialogue for the Replaces parameter on an attended transfer.", dialogue.Owner));
SIPResponse errorResponse = SIPTransport.GetResponse(referRequest, SIPResponseStatusCodesEnum.BadRequest, "Could not locate replaced dialogue");
referTransaction.SendFinalResponse(errorResponse);
}
else
{
logger.Debug("REFER dialogue being replaced " + replacesDialogue.DialogueName + ".");
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Replacement dialogue found on Refer, accepting.", dialogue.Owner));
bool sendNotifications = true;
if(!referRequest.Header.ReferSub.IsNullOrBlank())
{
Boolean.TryParse(referRequest.Header.ReferSub, out sendNotifications);
}
SIPDialogue remainingDialogue = GetOppositeDialogue(replacesDialogue);
SIPDialogue remaining2Dialogue = GetOppositeDialogue(dialogue);
logger.Debug("REFER dialogue remaining " + remainingDialogue.DialogueName + ".");
Guid newBridgeId = Guid.NewGuid();
remainingDialogue.BridgeId = newBridgeId;
remainingDialogue.CSeq++;
remaining2Dialogue.BridgeId = newBridgeId;
remaining2Dialogue.CSeq++;
m_sipDialoguePersistor.Update(new SIPDialogueAsset(remainingDialogue));
m_sipDialoguePersistor.Update(new SIPDialogueAsset(remaining2Dialogue));
Log_External(new SIPMonitorMachineEvent(SIPMonitorMachineEventTypesEnum.SIPDialogueUpdated, remainingDialogue.Owner, remainingDialogue.Id.ToString(), remainingDialogue.LocalUserField.URI));
Log_External(new SIPMonitorMachineEvent(SIPMonitorMachineEventTypesEnum.SIPDialogueUpdated, remaining2Dialogue.Owner, remaining2Dialogue.Id.ToString(), remaining2Dialogue.LocalUserField.URI));
Log_External(new SIPMonitorMachineEvent(SIPMonitorMachineEventTypesEnum.SIPDialogueTransfer, remainingDialogue.Owner, remainingDialogue.Id.ToString(), remainingDialogue.LocalUserField.URI));
SIPResponse acceptedResponse = SIPTransport.GetResponse(referRequest, SIPResponseStatusCodesEnum.Accepted, null);
referTransaction.SendFinalResponse(acceptedResponse);
if (sendNotifications)
{
SendNotifyRequestForRefer(referRequest, dialogue, localEndPoint, SIPResponseStatusCodesEnum.Trying, null);
}
logger.Debug("Reinviting " + remainingDialogue.DialogueName + " with " + remaining2Dialogue.DialogueName + ".");
ReInvite(remainingDialogue, remaining2Dialogue);
ReInvite(remaining2Dialogue, remainingDialogue);
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Transfer dialogue re-invites complete.", dialogue.Owner));
if (sendNotifications)
{
SendNotifyRequestForRefer(referRequest, dialogue, localEndPoint, SIPResponseStatusCodesEnum.Ok, null);
}
// Hangup redundant dialogues.
logger.Debug("Hanging up redundant dialogues post transfer.");
logger.Debug("Hanging up " + dialogue.DialogueName + ".");
dialogue.Hangup(m_sipTransport, m_outboundProxy);
CallHungup(dialogue, "Attended transfer", false);
logger.Debug("Hanging up " + replacesDialogue.DialogueName + ".");
replacesDialogue.Hangup(m_sipTransport, m_outboundProxy);
CallHungup(replacesDialogue, "Attended transfer", false);
}
}
}
catch (Exception excp)
{
logger.Error("Exception ProcessAttendedRefer. " + excp.Message);
throw;
}
}
/// <summary>
/// Establishes a new call with the client end tied to the proxy. Since the proxy will not be sending any audio the idea is that once
/// the call is up it should be re-INVITED off somewhere else pronto to avoid the callee sitting their listening to dead air.
/// </summary>
/// <param name="dest1">The dial string of the first call to place.</param>
/// <param name="dest2">The dial string of the second call to place.</param>
/// <param name="delaySeconds">Delay in seconds before placing the first call. Gives the user a chance to hangup their phone if they are calling themselves back.</param>
/// <param name="ringTimeoutLeg1">The ring timeout for the first call leg, If 0 the max timeout will be used.</param>
/// <param name="ringTimeoutLeg1">The ring timeout for the second call leg, If 0 the max timeout will be used.</param>
/// <param name="customHeadersCallLeg1">A | delimited string that contains a list of custom SIP headers to add to the INVITE request sent for the first call leg.</param>
/// /// <param name="customHeadersCallLeg2">A | delimited string that contains a list of custom SIP headers to add to the INVITE request sent for the second call leg.</param>
/// <returns>The result of the call.</returns>
public void Callback(string dest1, string dest2, int delaySeconds, int ringTimeoutLeg1, int ringTimeoutLeg2, string customHeadersCallLeg1, string customHeadersCallLeg2)
{
try
{
if (delaySeconds > 0)
{
delaySeconds = (delaySeconds > MAXCALLBACK_DELAY_SECONDS) ? MAXCALLBACK_DELAY_SECONDS : delaySeconds;
Log("Callback app delaying by " + delaySeconds + "s.");
Thread.Sleep(delaySeconds * 1000);
}
Log("Callback app commencing first leg to " + dest1 + ".");
SIPEndPoint defaultUDPEP = m_sipTransport.GetDefaultSIPEndPoint(SIPProtocolsEnum.udp);
SIPRequest firstLegDummyInviteRequest = GetCallbackInviteRequest(defaultUDPEP.GetIPEndPoint(), null);
ringTimeoutLeg1 = (ringTimeoutLeg1 > 0) ? ringTimeoutLeg1 : MAXCALLBACK_RINGTIME_SECONDS;
m_firstLegDialogue = Dial(dest1, ringTimeoutLeg1, 0, firstLegDummyInviteRequest, SIPCallDescriptor.ParseCustomHeaders(customHeadersCallLeg1));
if (m_firstLegDialogue == null)
{
Log("The first call leg to " + dest1 + " was unsuccessful.");
return;
}
SDP firstLegSDP = SDP.ParseSDPDescription(m_firstLegDialogue.RemoteSDP);
string call1SDPIPAddress = firstLegSDP.Connection.ConnectionAddress;
int call1SDPPort = firstLegSDP.Media[0].Port;
Log("The first call leg to " + dest1 + " was successful, audio socket=" + call1SDPIPAddress + ":" + call1SDPPort + ".");
Log("Callback app commencing second leg to " + dest2 + ".");
SIPRequest secondLegDummyInviteRequest = GetCallbackInviteRequest(defaultUDPEP.GetIPEndPoint(), m_firstLegDialogue.RemoteSDP);
ringTimeoutLeg2 = (ringTimeoutLeg2 > 0) ? ringTimeoutLeg2 : MAXCALLBACK_RINGTIME_SECONDS;
SIPDialogue secondLegDialogue = Dial(dest2, ringTimeoutLeg2, 0, secondLegDummyInviteRequest, SIPCallDescriptor.ParseCustomHeaders(customHeadersCallLeg2));
if (secondLegDialogue == null)
{
Log("The second call leg to " + dest2 + " was unsuccessful.");
m_firstLegDialogue.Hangup(m_sipTransport, m_outboundProxy);
return;
}
SDP secondLegSDP = SDP.ParseSDPDescription(secondLegDialogue.RemoteSDP);
string call2SDPIPAddress = secondLegSDP.Connection.ConnectionAddress;
int call2SDPPort = secondLegSDP.Media[0].Port;
Log("The second call leg to " + dest2 + " was successful, audio socket=" + call2SDPIPAddress + ":" + call2SDPPort + ".");
m_callManager.CreateDialogueBridge(m_firstLegDialogue, secondLegDialogue, m_username);
Log("Re-inviting Callback dialogues to each other.");
m_callManager.ReInvite(m_firstLegDialogue, secondLegDialogue);
//m_callManager.ReInvite(secondLegDialogue, m_firstLegDialogue.RemoteSDP);
SendRTPPacket(call2SDPIPAddress + ":" + call2SDPPort, call1SDPIPAddress + ":" + call1SDPPort);
SendRTPPacket(call1SDPIPAddress + ":" + call1SDPPort, call2SDPIPAddress + ":" + call2SDPPort);
}
catch (Exception excp)
{
logger.Error("Exception CallbackApp. " + excp);
Log("Exception in Callback. " + excp);
}
}
private void HangupDialogue(SIPDialogue dialogue, string hangupCause, bool sendBye)
{
try
{
//logger.Debug("Hanging up orphaned dialogue " + dialogue.DialogueName + ".");
if (dialogue.CDRId != Guid.Empty)
{
SIPCDRAsset cdr = m_sipCDRPersistor.Get(dialogue.CDRId);
if (cdr != null)
{
cdr.BridgeId = dialogue.BridgeId.ToString();
cdr.Hungup(hangupCause);
}
else
{
logger.Warn("CDR could not be found for remote dialogue in SIPCallManager CallHungup.");
}
}
else
{
logger.Warn("There was no CDR attached to orphaned dialogue in SIPCallManager CallHungup.");
}
if (sendBye)
{
dialogue.Hangup(m_sipTransport, m_outboundProxy);
}
m_sipDialoguePersistor.Delete(new SIPDialogueAsset(dialogue));
SIPEndPoint orphanedDialogueRemoteEP = (IPSocket.IsIPSocket(dialogue.RemoteTarget.Host)) ? SIPEndPoint.ParseSIPEndPoint(dialogue.RemoteTarget.Host) : null;
Log_External(new SIPMonitorMachineEvent(SIPMonitorMachineEventTypesEnum.SIPDialogueRemoved, dialogue.Owner, dialogue.Id.ToString(), dialogue.LocalUserField.URI));
}
catch (Exception excp)
{
logger.Error("Exception HangupDialogue. " + excp.Message);
}
}
/// <summary>
/// Establishes a new call with the client end tied to the proxy. Since the proxy will not be sending any audio the idea is that once
/// the call is up it should be re-INVITED off somewhere else pronto to avoid the callee sitting their listening to dead air.
/// </summary>
/// <param name="dest1">The dial string of the first call to place.</param>
/// <param name="dest2">The dial string of the second call to place.</param>
/// <param name="delaySeconds">Delay in seconds before placing the first call. Gives the user a chance to hangup their phone if they are calling themselves back.</param>
/// <param name="ringTimeoutLeg1">The ring timeout for the first call leg, If 0 the max timeout will be used.</param>
/// <param name="ringTimeoutLeg1">The ring timeout for the second call leg, If 0 the max timeout will be used.</param>
/// <param name="customHeadersCallLeg1">A | delimited string that contains a list of custom SIP headers to add to the INVITE request sent for the first call leg.</param>
/// /// <param name="customHeadersCallLeg2">A | delimited string that contains a list of custom SIP headers to add to the INVITE request sent for the second call leg.</param>
/// <returns>The result of the call.</returns>
public void Callback(string dest1, string dest2, int delaySeconds, int ringTimeoutLeg1, int ringTimeoutLeg2, string customHeadersCallLeg1, string customHeadersCallLeg2)
{
var ts = new CancellationTokenSource();
CancellationToken ct = ts.Token;
try
{
if (delaySeconds > 0)
{
delaySeconds = (delaySeconds > MAXCALLBACK_DELAY_SECONDS) ? MAXCALLBACK_DELAY_SECONDS : delaySeconds;
Log("Callback app delaying by " + delaySeconds + "s.");
Thread.Sleep(delaySeconds * 1000);
}
Log("Callback app commencing first leg to " + dest1 + ".");
SIPEndPoint defaultUDPEP = m_sipTransport.GetDefaultSIPEndPoint(SIPProtocolsEnum.udp);
SIPRequest firstLegDummyInviteRequest = GetCallbackInviteRequest(defaultUDPEP.GetIPEndPoint(), null);
ForkCall firstLegCall = new ForkCall(m_sipTransport, Log_External, m_callManager.QueueNewCall, null, m_username, m_adminMemberId, m_outboundProxy, m_callManager, null);
m_firstLegDialogue = Dial(firstLegCall, dest1, ringTimeoutLeg1, 0, firstLegDummyInviteRequest, SIPCallDescriptor.ParseCustomHeaders(customHeadersCallLeg1));
if (m_firstLegDialogue == null)
{
Log("The first call leg to " + dest1 + " was unsuccessful.");
return;
}
// Persist the dialogue to the database so any hangup can be detected.
m_sipDialoguePersistor.Add(new SIPDialogueAsset(m_firstLegDialogue));
SDP firstLegSDP = SDP.ParseSDPDescription(m_firstLegDialogue.RemoteSDP);
string call1SDPIPAddress = firstLegSDP.Connection.ConnectionAddress;
int call1SDPPort = firstLegSDP.Media[0].Port;
Log("The first call leg to " + dest1 + " was successful, audio socket=" + call1SDPIPAddress + ":" + call1SDPPort + ".");
Log("Callback app commencing second leg to " + dest2 + ".");
SIPRequest secondLegDummyInviteRequest = GetCallbackInviteRequest(defaultUDPEP.GetIPEndPoint(), m_firstLegDialogue.RemoteSDP);
ForkCall secondLegCall = new ForkCall(m_sipTransport, Log_External, m_callManager.QueueNewCall, null, m_username, m_adminMemberId, m_outboundProxy, m_callManager, null);
Task.Factory.StartNew(() =>
{
while (true)
{
Thread.Sleep(CHECK_FIRST_LEG_FOR_HANGUP_PERIOD);
Console.WriteLine("Checking if first call leg is still up...");
if (ct.IsCancellationRequested)
{
Console.WriteLine("Checking first call leg task was cancelled.");
break;
}
else
{
// Check that the first call leg hasn't been hung up.
var dialog = m_sipDialoguePersistor.Get(m_firstLegDialogue.Id);
if (dialog == null)
{
Console.WriteLine("First call leg has been hungup.");
// The first call leg has been hungup while waiting for the second call.
Log("The first call leg was hungup while the second call leg was waiting for an answer.");
secondLegCall.CancelNotRequiredCallLegs(CallCancelCause.ClientCancelled);
break;
}
}
}
Console.WriteLine("Checking first call leg task finished...");
}, ct);
SIPDialogue secondLegDialogue = Dial(secondLegCall, dest2, ringTimeoutLeg2, 0, secondLegDummyInviteRequest, SIPCallDescriptor.ParseCustomHeaders(customHeadersCallLeg2));
ts.Cancel();
if (secondLegDialogue == null)
{
Log("The second call leg to " + dest2 + " was unsuccessful.");
m_firstLegDialogue.Hangup(m_sipTransport, m_outboundProxy);
return;
}
// Check that the first call leg hasn't been hung up.
var firstLegDialog = m_sipDialoguePersistor.Get(m_firstLegDialogue.Id);
if (firstLegDialog == null)
{
// The first call leg has been hungup while waiting for the second call.
Log("The first call leg was hungup while waiting for the second call leg.");
secondLegDialogue.Hangup(m_sipTransport, m_outboundProxy);
return;
}
SDP secondLegSDP = SDP.ParseSDPDescription(secondLegDialogue.RemoteSDP);
string call2SDPIPAddress = secondLegSDP.Connection.ConnectionAddress;
int call2SDPPort = secondLegSDP.Media[0].Port;
Log("The second call leg to " + dest2 + " was successful, audio socket=" + call2SDPIPAddress + ":" + call2SDPPort + ".");
// Persist the second leg dialogue and update the bridge ID on the first call leg.
Guid bridgeId = Guid.NewGuid();
secondLegDialogue.BridgeId = bridgeId;
m_sipDialoguePersistor.Add(new SIPDialogueAsset(secondLegDialogue));
m_sipDialoguePersistor.UpdateProperty(firstLegDialog.Id, "BridgeID", bridgeId.ToString());
//m_callManager.CreateDialogueBridge(m_firstLegDialogue, secondLegDialogue, m_username);
Log("Re-inviting Callback dialogues to each other.");
m_callManager.ReInvite(m_firstLegDialogue, secondLegDialogue);
//m_callManager.ReInvite(secondLegDialogue, m_firstLegDialogue.RemoteSDP);
SendRTPPacket(call2SDPIPAddress + ":" + call2SDPPort, call1SDPIPAddress + ":" + call1SDPPort);
SendRTPPacket(call1SDPIPAddress + ":" + call1SDPPort, call2SDPIPAddress + ":" + call2SDPPort);
}
catch (Exception excp)
{
logger.Error("Exception CallbackApp. " + excp);
Log("Exception in Callback. " + excp);
}
finally
{
if (!ts.IsCancellationRequested)
{
ts.Cancel();
}
}
}
