[asterisk-scf-commits] asterisk-scf/integration/sip.git branch "master" updated.

[Commits to the Asterisk SCF project code repositories]

branch "master" has been updated
       via  33cb866433a631c952b3c17a668da144fc4a7705 (commit)
      from  b1894c18bc03ee869333c5a77cecaa2f0da3ddcf (commit)

Summary of changes:
 slice                      |    2 +-
 src/PJSipSessionModule.cpp |   63 ++++++++++++++++++++++++++++++++++++++++++-
 2 files changed, 62 insertions(+), 3 deletions(-)


- Log -----------------------------------------------------------------
commit 33cb866433a631c952b3c17a668da144fc4a7705
Author: Mark Michelson <mmichelson at digium.com>
Date:   Wed Aug 25 10:06:47 2010 -0500

    Create INVITE session UAS on reception of INVITE.
    
    Also now will send a 100 Trying message.

diff --git a/slice b/slice
index dcb271b..d6b2777 160000
--- a/slice
+++ b/slice
@@ -1 +1 @@
-Subproject commit dcb271baaca90fa89ed6b7d4846ea458fb303943
+Subproject commit d6b2777545b3b8eae7c5dc9b40a1ca3ff9e26a43
diff --git a/src/PJSipSessionModule.cpp b/src/PJSipSessionModule.cpp
index de09e0f..0ca0ff8 100644
--- a/src/PJSipSessionModule.cpp
+++ b/src/PJSipSessionModule.cpp
@@ -54,8 +54,54 @@ static void handle_new_invite(pjsip_rx_data *rdata)
 	//sending a 100 trying, finding the remote endpoint
 	//to call, and placing a call to it.
 	
+	//XXX All failure cases need to have cleanup and
+	//returing added in addition to printing a message
+	//as they do now.
+	
 	//XXX Put caller identification code in here!
 	
+	//XXX According to pjsip docs, we should call
+	//pjsip_inv_verify_request to be sure we can
+	//handle the request. For now, just plunge forward.
+	
+	pjsip_dialog *dlg;
+	//XXX The NULL parameter should be replaced with
+	//An appropriate Contact header. Leaving it NULL makes
+	//PJSIP create the contact header for responses based
+	//on the To header of the incoming Invite.
+	pj_status_t status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, NULL, &dlg);
+
+	if (status != PJ_SUCCESS)
+	{
+		std::cerr << "[WARNING] Unable to create UAS dialog on incoming INVITE" << std::endl;
+	}
+
+	//XXX The sdp argument is NULL for now, but can be changed if we
+	//know what has been configured for this particular caller.
+	
+	pjsip_inv_session *inv_session;
+	status = pjsip_inv_create_uas(dlg, rdata, NULL, 0, &inv_session);
+
+	if (status != PJ_SUCCESS)
+	{
+		std::cerr << "[WARNING] Unable to create INVITE session" << std::endl;
+	}
+
+	pjsip_tx_data *tdata;
+	status = pjsip_inv_initial_answer(inv_session, rdata, 100, NULL, NULL, &tdata);
+
+	if (status != PJ_SUCCESS)
+	{
+		std::cerr << "[WARNING] Failed to create 100 Trying response" << std::endl;
+	}
+
+	status = pjsip_inv_send_msg(inv_session, tdata);
+
+	if (status != PJ_SUCCESS)
+	{
+		std::cerr << "[WARNING] Failed to send 100 Trying response" << std::endl;
+	}
+
 	SipChannelServiceDataModel &dataModel = SipChannelServiceDataModel::getInstance();
 	boost::shared_ptr<SipEndpointFactory> factory = dataModel.getEndpointFactory();
 	//XXX Hardcoded "Butt" is bad for a lot of reasons,
@@ -78,11 +124,11 @@ static void handle_new_invite(pjsip_rx_data *rdata)
 	{
 		endpoints = locator->lookup(destination);
 	}
-	catch (DestinationNotFoundException destEx)
+	catch (DestinationNotFoundException &destEx)
 	{
 		std::cerr << "[WARNING] Could not find destination " << destEx.destination << std::endl;
 	}
-	catch (InvalidParamsException invalEx)
+	catch (InvalidParamsException &invalEx)
 	{
 		std::cerr << "[WARNING] Invalid parameters in endpoint lookup. " << std::endl;
 	}
@@ -263,6 +309,12 @@ static pjsip_redirect_op invOnRedirected(pjsip_inv_session *inv, const pjsip_uri
 	return PJSIP_REDIRECT_REJECT;
 }
 
+static pjsip_dialog *uaOnDialogForked(pjsip_dialog *first_set, pjsip_rx_data *rdata)
+{
+	//stub
+	return NULL;
+}
+
 PJSipSessionModule::PJSipSessionModule() : mName("Session Module")
 {
 	// XXX Set the module name here. It's a pain in the neck
@@ -280,6 +332,12 @@ PJSipSessionModule::PJSipSessionModule() : mName("Session Module")
 	mModule.on_tx_response = sessionOnTransmitResponse;
 	mModule.on_tsx_state = sessionOnTransactionStateChange;
 
+	if (pjsip_ua_instance()->id == -1)
+	{
+		pj_bzero(&mUaParam, sizeof(&mUaParam));
+		mUaParam.on_dlg_forked = uaOnDialogForked;
+	}
+
 	if (pjsip_inv_usage_instance()->id == -1)
 	{
 		pj_bzero(&mInvCallback, sizeof(&mInvCallback));
@@ -295,6 +353,7 @@ PJSipSessionModule::PJSipSessionModule() : mName("Session Module")
 	
 	PJSipManager *manager = PJSipManager::getInstance();
 	pjsip_endpoint *endpt = manager->getEndpoint();
+	pjsip_ua_init_module(endpt, &mUaParam);
 	pjsip_inv_usage_init(endpt, &mInvCallback);
 	manager->registerModule(&mModule);
 }

-----------------------------------------------------------------------


-- 
asterisk-scf/integration/sip.git