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savoirfairelinux
jami-daemon
Commits
b1d1eb46
Commit
b1d1eb46
authored
Apr 03, 2008
by
Emmanuel Milou
Browse files
SIP transfer OK
The dtmf's was sent. Bad
parent
38d095a5
Changes
3
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Inline
Side-by-side
sflphone-gtk/src/actions.c
View file @
b1d1eb46
...
...
@@ -55,6 +55,28 @@ sflphone_notify_voice_mail (guint count)
status_bar_message_add
(
message
,
__MSG_VOICE_MAILS
);
g_free
(
message
);
}
// TODO: add ifdef
if
(
account_list_get_size
()
>
0
)
{
account_t
*
acc
=
account_list_get_by_state
(
ACCOUNT_STATE_REGISTERED
);
if
(
acc
==
NULL
)
{
// Notify that no account is registered
//notify_no_account_registered();
}
else
{
if
(
account_list_get_default
()
==
NULL
){
// Notify that the first registered account has count voice mails
notify_voice_mails
(
count
,
acc
);
}
else
{
// Notify that the default registered account has count voice mails
notify_voice_mails
(
count
,
account_list_get_by_id
(
account_list_get_default
())
);
}
}
}
}
void
...
...
@@ -345,6 +367,7 @@ sflphone_unset_transfert()
}
toolbar_update_buttons
();
}
void
sflphone_incoming_call
(
call_t
*
c
)
{
...
...
@@ -357,7 +380,7 @@ sflphone_incoming_call (call_t * c)
void
process_dialing
(
call_t
*
c
,
guint
keyval
,
gchar
*
key
)
{
// We stop the tone
if
(
strlen
(
c
->
to
)
==
0
){
if
(
strlen
(
c
->
to
)
==
0
&&
c
->
state
!=
CALL_STATE_TRANSFERT
){
dbus_start_tone
(
FALSE
,
0
);
dbus_play_dtmf
(
key
);
}
...
...
@@ -421,10 +444,10 @@ void process_dialing(call_t * c, guint keyval, gchar * key)
call_t
*
sflphone_new_call
()
{
// Play a tone when creating a new call
if
(
call_list_get_size
()
==
0
)
{
dbus_start_tone
(
TRUE
,
(
voice_mails
>
0
)
?
TONE_WITH_MESSAGE
:
TONE_WITHOUT_MESSAGE
)
;
}
call_t
*
c
=
g_new0
(
call_t
,
1
);
c
->
state
=
CALL_STATE_DIALING
;
c
->
from
=
g_strconcat
(
"
\"\"
<>"
,
NULL
);
...
...
src/managerimpl.cpp
View file @
b1d1eb46
...
...
@@ -377,6 +377,7 @@ ManagerImpl::transferCall(const CallID& id, const std::string& to)
if
(
_dbus
)
_dbus
->
getCallManager
()
->
callStateChanged
(
id
,
"HUNGUP"
);
switchCall
(
""
);
return
returnValue
;
}
...
...
src/sipvoiplink.cpp
View file @
b1d1eb46
...
...
@@ -840,6 +840,7 @@ SIPVoIPLink::transfer(const CallID& id, const std::string& to)
}
eXosip_unlock
();
_audiortp
.
closeRtpSession
();
// shall we delete the call?
//removeCall(id);
return
true
;
...
...
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