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Call/SipCall pointer to enable jerome to compile

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This commit is contained in:
yanmorin
2005-10-05 21:51:58 +00:00
parent 1e1ea7960c
commit 702b844520
2 changed files with 8 additions and 7 deletions
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2
src/config/config.cpp
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@ -64,7 +64,7 @@ ConfigTree::addConfigTreeItem(const std::string& section, const ConfigTreeItem i
iter = _sections.find(section);
}
// be prudent here
if (iter!=NULL && iter != _sections.end()) {
if (iter != NULL && iter != _sections.end()) {
std::string name = item.getName();
if ( iter->second->find(name) == iter->second->end()) {

13
src/sipvoiplink.cpp
View File

@ -550,6 +550,8 @@ SipVoIPLink::getEvent (void)
eXosip_event_t *event;
short id;
char *name;
Call *call = NULL;
SipCall *sipcall = NULL;
event = eXosip_event_wait (0, 50);
eXosip_lock();
@ -583,7 +585,7 @@ SipVoIPLink::getEvent (void)
// Generate id
id = Manager::instance().generateNewCallId();
Manager::instance().pushBackNewCall(id, Incoming);
call = Manager::instance().pushBackNewCall(id, Incoming);
_debug("Incoming Call with id %d [cid = %d, did = %d]\n",
id, event->cid, event->did);
_debug("Local audio port: %d\n", _localPort);
@ -592,22 +594,22 @@ SipVoIPLink::getEvent (void)
osip_from_t *from;
osip_from_init(&from);
sipcall = getSipCall(id);
if (event->request != NULL) {
char *tmp = NULL;
osip_from_to_str (event->request->from, &tmp);
if (tmp != NULL) {
snprintf (getSipCall(id)->getRemoteUri(), 256, "%s", tmp);
snprintf (sipcall->getRemoteUri(), 256, "%s", tmp);
osip_free (tmp);
}
}
osip_from_parse(from, getSipCall(id)->getRemoteUri());
osip_from_parse(from, sipcall->getRemoteUri());
name = osip_from_get_displayname(from);
//Don't need this display text message now that we send the name
//inside the Manager to the gui
//Manager::instance().displayTextMessage(id, name);
Call *call = Manager::instance().getCall(id);
if ( call != NULL) {
call->setCallerIdName(name);
osip_uri_t* url = osip_from_get_url(from);
@ -622,7 +624,6 @@ SipVoIPLink::getEvent (void)
_debug("From: %s\n", name);
osip_from_free(from);
SipCall *sipcall = getSipCall(id);
// Associate an audio port with a call
sipcall->setLocalAudioPort(_localPort);
sipcall->setLocalIp(getLocalIpAddress());
@ -648,7 +649,7 @@ SipVoIPLink::getEvent (void)
if (id == 0) {
id = findCallIdInitial(event);
}
SipCall *sipcall = getSipCall(id);
sipcall = getSipCall(id);
if ( sipcall ) {
_debug("Call is answered [id = %d, cid = %d, did = %d], localport=%d\n",
id, event->cid, event->did,sipcall->getLocalAudioPort());