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savoirfairelinux
jami-daemon
Commits
97c84363
Commit
97c84363
authored
Feb 03, 2009
by
Emmanuel Milou
Browse files
Function terminateSIPCall added in sipvoiplink and managerimpl
Conflicts: src/audio/audiortp.cpp src/sipvoiplink.cpp
parent
48e1e942
Changes
3
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Inline
Side-by-side
src/audio/audiortp.cpp
View file @
97c84363
...
...
@@ -126,7 +126,6 @@ AudioRtpRTX::~AudioRtpRTX () {
_debugException
(
"! ARTP: Thread destructor didn't terminate correctly"
);
throw
;
}
//_debug("terminate audiortprtx ended...\n");
_ca
=
0
;
if
(
!
_sym
)
{
delete
_sessionRecv
;
_sessionRecv
=
NULL
;
...
...
@@ -145,6 +144,7 @@ AudioRtpRTX::~AudioRtpRTX () {
delete
time
;
time
=
NULL
;
delete
converter
;
converter
=
NULL
;
}
void
...
...
src/managerimpl.cpp
View file @
97c84363
...
...
@@ -249,6 +249,7 @@ ManagerImpl::answerCall(const CallID& id)
bool
ManagerImpl
::
hangupCall
(
const
CallID
&
id
)
{
_debug
(
"ManagerImpl::hangupCall(): This function is called when user hangup
\n
"
);
PulseLayer
*
pulselayer
;
AccountID
accountid
;
bool
returnValue
;
...
...
@@ -370,6 +371,7 @@ ManagerImpl::transferCall(const CallID& id, const std::string& to)
bool
ManagerImpl
::
refuseCall
(
const
CallID
&
id
)
{
_debug
(
"ManagerImpl::refuseCall(): method called"
);
stopTone
(
true
);
AccountID
accountid
=
getAccountFromCall
(
id
);
if
(
accountid
==
AccountNULL
)
{
...
...
@@ -632,8 +634,10 @@ ManagerImpl::peerRingingCall(const CallID& id)
void
ManagerImpl
::
peerHungupCall
(
const
CallID
&
id
)
{
_debug
(
"ManagerImpl::peerHungupCall():this function is called when peer hangup
\n
"
);
PulseLayer
*
pulselayer
;
AccountID
accountid
;
bool
returnValue
;
accountid
=
getAccountFromCall
(
id
);
if
(
accountid
==
AccountNULL
)
{
...
...
@@ -649,6 +653,8 @@ ManagerImpl::peerHungupCall(const CallID& id)
switchCall
(
""
);
}
returnValue
=
getAccountLink
(
accountid
)
->
hangup
(
id
);
removeWaitingCall
(
id
);
removeCallAccount
(
id
);
...
...
src/sipvoiplink.cpp
View file @
97c84363
...
...
@@ -149,6 +149,8 @@ SIPVoIPLink* SIPVoIPLink::_instance = NULL;
,
_useStun
(
false
)
,
_clients
(
0
)
{
_debug
(
"SIPVoIPLink::~SIPVoIPLink(): sipvoiplink constructor called
\n
"
);
// to get random number for RANDOM_PORT
srand
(
time
(
NULL
));
...
...
@@ -158,6 +160,7 @@ SIPVoIPLink* SIPVoIPLink::_instance = NULL;
SIPVoIPLink
::~
SIPVoIPLink
()
{
_debug
(
"SIPVoIPLink::~SIPVoIPLink(): sipvoiplink destructor called
\n
"
);
terminate
();
}
...
...
@@ -213,7 +216,6 @@ SIPVoIPLink::terminate()
void
SIPVoIPLink
::
terminateSIPCall
()
{
ost
::
MutexLock
m
(
_callMapMutex
);
CallMap
::
iterator
iter
=
_callMap
.
begin
();
SIPCall
*
call
;
...
...
@@ -488,6 +490,8 @@ SIPVoIPLink::hangup(const CallID& id)
_debug
(
"* SIP Info: Stopping AudioRTP for hangup
\n
"
);
_audiortp
->
closeRtpSession
();
}
terminateSIPCall
();
removeCall
(
id
);
...
...
@@ -689,7 +693,7 @@ SIPVoIPLink::refuse (const CallID& id)
pj_status_t
status
;
pjsip_tx_data
*
tdata
;
_debug
(
"SIPVoIPLink::refuse() : teh call is refused
\n
"
);
call
=
getSIPCall
(
id
);
if
(
call
==
0
)
{
...
...
@@ -713,6 +717,8 @@ SIPVoIPLink::refuse (const CallID& id)
return
false
;
call
->
getInvSession
()
->
mod_data
[
getModId
()]
=
NULL
;
terminateSIPCall
();
return
true
;
}
...
...
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